<pre>Network Working Group                                           S. Floyd
Request for Comments: 4336                                          ICIR
Category: Informational                                       M. Handley
                                                                     UCL
                                                               E. Kohler
                                                                    UCLA
                                                              March 2006


                       <span class="h1">Problem Statement for the</span>
              <span class="h1">Datagram Congestion Control Protocol (DCCP)</span>

Status of This Memo

   This memo provides information for the Internet community.  It does
   not specify an Internet standard of any kind.  Distribution of this
   memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   This document describes for the historical record the motivation
   behind the Datagram Congestion Control Protocol (DCCP), an unreliable
   transport protocol incorporating end-to-end congestion control.  DCCP
   implements a congestion-controlled, unreliable flow of datagrams for
   use by applications such as streaming media or on-line games.






















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Table of Contents

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-2">2</a>
   <a href="#section-2">2</a>. Problem Space ...................................................<a href="#page-3">3</a>
      <a href="#section-2.1">2.1</a>. Congestion Control for Unreliable Transfer .................<a href="#page-4">4</a>
      <a href="#section-2.2">2.2</a>. Overhead ...................................................<a href="#page-6">6</a>
      <a href="#section-2.3">2.3</a>. Firewall Traversal .........................................<a href="#page-6">6</a>
      <a href="#section-2.4">2.4</a>. Parameter Negotiation ......................................<a href="#page-7">7</a>
   <a href="#section-3">3</a>. Solution Space for Congestion Control of Unreliable Flows .......<a href="#page-7">7</a>
      <a href="#section-3.1">3.1</a>. Providing Congestion Control Above UDP .....................<a href="#page-8">8</a>
           <a href="#section-3.1.1">3.1.1</a>. The Burden on the Application Designer ..............<a href="#page-8">8</a>
           <a href="#section-3.1.2">3.1.2</a>. Difficulties with ECN ...............................<a href="#page-8">8</a>
           <a href="#section-3.1.3">3.1.3</a>. The Evasion of Congestion Control ..................<a href="#page-10">10</a>
      <a href="#section-3.2">3.2</a>. Providing Congestion Control Below UDP ....................<a href="#page-10">10</a>
           <a href="#section-3.2.1">3.2.1</a>. Case 1: Congestion Feedback at the Application .....<a href="#page-11">11</a>
           <a href="#section-3.2.2">3.2.2</a>. Case 2: Congestion Feedback at a Layer Below UDP ...<a href="#page-11">11</a>
      <a href="#section-3.3">3.3</a>. Providing Congestion Control at the Transport Layer .......<a href="#page-12">12</a>
           <a href="#section-3.3.1">3.3.1</a>. Modifying TCP? .....................................<a href="#page-12">12</a>
           <a href="#section-3.3.2">3.3.2</a>. Unreliable Variants of SCTP? .......................<a href="#page-13">13</a>
           <a href="#section-3.3.3">3.3.3</a>. Modifying RTP? .....................................<a href="#page-14">14</a>
           <a href="#section-3.3.4">3.3.4</a>. Designing a New Transport Protocol .................<a href="#page-14">14</a>
   <a href="#section-4">4</a>. Selling Congestion Control to Reluctant Applications ...........<a href="#page-15">15</a>
   <a href="#section-5">5</a>. Additional Design Considerations ...............................<a href="#page-15">15</a>
   <a href="#section-6">6</a>. Transport Requirements of Request/Response Applications ........<a href="#page-16">16</a>
   <a href="#section-7">7</a>. Summary of Recommendations .....................................<a href="#page-17">17</a>
   <a href="#section-8">8</a>. Security Considerations ........................................<a href="#page-18">18</a>
   <a href="#section-9">9</a>. Acknowledgements ...............................................<a href="#page-18">18</a>
   Informative References ............................................<a href="#page-19">19</a>

<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   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 streaming audio,
   Internet telephony, and multiplayer and massively multiplayer on-line
   games, 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



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   congestion-controlled traffic, poses a real threat to the overall
   health of the Internet [RFC2914, <a href="./rfc3714">RFC3714</a>].

   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 will 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 describe why a new protocol is the best
   solution for the more general problem of bringing congestion control
   to unreliable flows of unicast datagrams, and discuss briefly
   subsidiary requirements for mobility, defense against Denial of
   Service (DoS) attacks and spoofing, interoperation with RTP, and
   interactions with Network Address Translators (NATs) and firewalls.

   One of the design preferences that we bring to this question is a
   preference for a clean, understandable, low-overhead, and minimal
   protocol.  As described later in this document, this results in the
   design decision to leave functionality such as reliability or Forward
   Error Correction (FEC) to be layered on top, rather than provided in
   the transport protocol itself.

   This document began in 2002 as a formalization of the goals of DCCP,
   the Datagram Congestion Control Protocol [<a href="./rfc4340" title="&quot;Datagram Congestion Control Protocol (DCCP)&quot;">RFC4340</a>].  We intended DCCP
   to satisfy this problem statement, and thus the original reasoning
   behind many of DCCP's design choices can be found here.  However, we
   believed, and continue to believe, that the problem should be solved
   whether or not DCCP is the chosen solution.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Problem Space</span>

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

   o  The potential for non-congestion-controlled datagram flows to
      cause congestion collapse of the network.  (See <a href="./rfc2914#section-5">Section&nbsp;5 of
      [RFC2914]</a> and <a href="./rfc3714#section-2">Section&nbsp;2 of [RFC3714]</a>.)







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

   o  The lack of a standard solution for reliably transmitting
      congestion feedback for an unreliable data flow.

   o  The lack of a standard solution for negotiating Explicit
      Congestion Notification (ECN) [<a href="./rfc3168" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>] usage for unreliable
      flows.

   o  The lack of a choice of TCP-friendly congestion control
      mechanisms.

   We assume that most application writers would use congestion control
   for long-lived unreliable flows if it were available in a standard,
   easy-to-use form.

   More minor issues include the following:

   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.

<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>.  Congestion Control for Unreliable Transfer</span>

   We aim to bring easy-to-use congestion control mechanisms to
   applications that generate large or long-lived flows of unreliable
   datagrams, such as RealAudio, Internet telephony, and multiplayer
   games.  Our motivation is to avoid congestion collapse.  (The short
   flows generated by request-response applications, such as DNS and
   SNMP, 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 require.





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

   o  Startup Delay: they wish to avoid the delay of a three-way
      handshake before initiating data transfer.

   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 issues, applications that generate large or long-lived flows
   of datagrams, such as media transfer and games, mostly care about
   controlling the trade-off 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
      playout.

   o  In an interactive game or virtual-reality 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.

   In a congestion-controlled flow, the allowed packet sending rate
   depends on measured network congestion.  Thus, some control is given
   up 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



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   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, TCP-Friendly Rate Control (TFRC)
   [<a href="./rfc3448" title="&quot;TCP Friendly Rate Control (TFRC): Protocol Specification&quot;">RFC3448</a>] rather than strict TCP-like congestion control.  (TFRC has
   been standardized in the IETF as a congestion control mechanism that
   adjusts its sending rate more smoothly than TCP does, while
   maintaining long-term fair bandwidth sharing with TCP [<a href="./rfc3448" title="&quot;TCP Friendly Rate Control (TFRC): Protocol Specification&quot;">RFC3448</a>].)

<span class="h3"><a class="selflink" id="section-2.2" href="#section-2.2">2.2</a>.  Overhead</span>

   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 5.6 Kbps data stream
   but wanting moderately low delay may send a packet every 20 ms,
   sending only 14 data bytes in each packet.  In addition, 20 bytes is
   taken up by the IP header, with additional bytes for transport and/or
   application headers.  Clearly, it is desirable for such an
   application to have a low-overhead transport protocol header.

   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
   datagrams.

<span class="h3"><a class="selflink" id="section-2.3" href="#section-2.3">2.3</a>.  Firewall Traversal</span>

   Applications requiring a flow of unreliable datagrams currently tend
   to use signalling protocols such as the Real Time Streaming Protocol
   (RTSP) [<a href="./rfc2326" title="&quot;Real Time Streaming Protocol (RTSP)&quot;">RFC2326</a>], SIP [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], 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 after being redirected or relayed to other machines.




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   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.  Although, for bi-directional flows, the
   firewall can open a bi-directional hole if it receives a UDP packet
   from inside the firewall, 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 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.

<span class="h3"><a class="selflink" id="section-2.4" href="#section-2.4">2.4</a>.  Parameter Negotiation</span>

   Different applications have different requirements for congestion
   control, which may map into different congestion feedback.  Examples
   include ECN capability and desired congestion control dynamics (the
   choice of congestion control algorithm and, therefore, the form of
   feedback information required).  Such parameters need to be reliably
   negotiated before congestion control can function correctly.

   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.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Solution Space for Congestion Control of Unreliable Flows</span>

   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
   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 in an alternative to
      UDP.



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   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.

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  Providing Congestion Control Above UDP</span>

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

<span class="h4"><a class="selflink" id="section-3.1.1" href="#section-3.1.1">3.1.1</a>.  The Burden on the Application Designer</span>

   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 [<a href="./rfc2525" title="&quot;Known TCP Implementation Problems&quot;">RFC2525</a>, <a href="#ref-PF01" title="&quot;Identifying the TCP Behavior of Web Servers&quot;">PF01</a>]
   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 <a href="#section-3.3.3">Section 3.3.3</a> for more
   discussion of RTP.)

   In addition to congestion control issues, we also note the problems
   with firewall traversal and parameter negotiation discussed in
   Sections <a href="#section-2.3">2.3</a> and <a href="#section-2.4">2.4</a>.  Implementing on top of UDP requires that the
   application designer also address these issues.

<span class="h4"><a class="selflink" id="section-3.1.2" href="#section-3.1.2">3.1.2</a>.  Difficulties with ECN</span>

   There is a second problem with providing congestion control above
   UDP: 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.




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   With the development of the ECN nonce, ECN can be useful even in the
   absence of network support.  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 ECN contract is that senders who set the ECT codepoint must
   respond to Congestion Experienced (CE) codepoints by reducing their
   sending rates.  Therefore, the ECT codepoint can only safely be set
   in the packet header of a UDP packet if the following is guaranteed:

   o  if the CE codepoint is set by a router, the receiving IP layer
      will pass the CE status to the UDP layer, which will pass it 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 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.

   In the absence of more fine-grained mechanisms for dealing with a
   period of sustained congestion, one possibility would be for routers
   to discontinue using ECN with UDP packets during the congested
   period, and to use ECN only with TCP or DCCP packets.  This would be
   a reasonable response, for example, if TCP or DCCP flows were found



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   to be more likely to be using conformant end-to-end congestion
   control than were UDP flows.  If routers were to adopt such a policy,
   then DCCP flows could be more likely to receive the benefits of ECN
   in times of congestion than would UDP flows.

<span class="h4"><a class="selflink" id="section-3.1.3" href="#section-3.1.3">3.1.3</a>.  The Evasion of Congestion Control</span>

   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.

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  Providing Congestion Control Below UDP</span>

   Instead of providing congestion control above UDP, a second
   possibility would be to provide congestion control for unreliable
   applications at a layer below UDP, with applications using 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:

   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 (CM) [<a href="./rfc3124" title="&quot;The Congestion Manager&quot;">RFC3124</a>].

   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 [<a href="#ref-BRS99" title="&quot;An Integrated Congestion Management Architecture for Internet Hosts&quot;">BRS99</a>] 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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<span class="h4"><a class="selflink" id="section-3.2.1" href="#section-3.2.1">3.2.1</a>.  Case 1: Congestion Feedback at the Application</span>

   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
   <a href="#section-3.1">Section 3.1</a>, 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.  Thus, this design limits the choice of
   congestion control mechanisms available to applications while
   simultaneously burdening the applications with implementations of
   congestion feedback.

<span class="h4"><a class="selflink" id="section-3.2.2" href="#section-3.2.2">3.2.2</a>.  Case 2: Congestion Feedback at a Layer Below UDP</span>

   Providing feedback at a layer below UDP would require an additional
   packet header below UDP to carry sequence numbers in addition to the
   8-byte header for UDP itself.  Unless this header were an IP option
   (which is likely to cause problems for many IPv4 routers), 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, and the modified protocol would face deployment challenges
   similar to those of an entirely new protocol.

   To use congestion feedback at a layer below UDP most effectively, the
   semantics of the UDP socket Application Programming Interface (API)
   would also need changing, both to support a late decision on what to
   send and to provide access to sequence numbers (so that the
   application wouldn't need to duplicate them for its own purposes).
   Thus, the socket API would no longer look like UDP to 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.






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<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  Providing Congestion Control at the Transport Layer</span>

   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 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 the Stream
   Control Transmission Protocol (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.

<span class="h4"><a class="selflink" id="section-3.3.1" href="#section-3.3.1">3.3.1</a>.  Modifying TCP?</span>

   One alternative might be to create an unreliable variant of TCP, with
   reliability layered on top for applications desiring reliable
   delivery.  However, our requirement is not simply for TCP minus 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, not datagrams.  This would complicate both congestion
   feedback and any attempt to allow the application to decide, at
   transmission time, what information should go into a packet.
   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.

   It seems best simply to leave TCP as it is, and to create a new
   congestion control protocol for unreliable transfer.  This is
   especially true since any change to TCP, no matter how small, takes
   an inordinate amount of time to standardize and deploy, given TCP's
   importance in the current Internet and the historical difficulty of
   getting TCP implementations right.




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<span class="h4"><a class="selflink" id="section-3.3.2" href="#section-3.3.2">3.3.2</a>.  Unreliable Variants of SCTP?</span>

   SCTP, the Stream Control Transmission Protocol [<a href="./rfc2960" title="&quot;Stream Control Transmission Protocol&quot;">RFC2960</a>], 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 size that accompany
   the use of streams in SCTP.

   The basic underlying structure of the SCTP packet, of 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 (ECNE) and Congestion
   Window Reduced (CWR) 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.

   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.

   Finally, the SCTP Partial Reliability extension [<a href="./rfc3758" title="&quot;Stream Control Transmission Protocol (SCTP) Partial Reliability Extension&quot;">RFC3758</a>] allows a
   sender to selectively abandon outstanding messages, which ceases
   retransmissions and allows the receiver to deliver any queued
   messages on the affected streams.  This service model, although
   well-suited for some applications, differs from, and provides the
   application somewhat less flexibility than, UDP's fully unreliable
   service.

   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.
   We would argue against this.  SCTP is well-suited for applications
   that desire limited retransmission with multistream or multihoming
   support.  Adding support for fully-unreliable variants, multiple
   congestion control profiles, and reduced single-stream headers would
   risk introducing unforeseen interactions and make further



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   modifications ever more difficult.  We have chosen instead to
   implement a minimal protocol, designed for fully-unreliable datagram
   service, that provides only end-to-end congestion control and any
   other mechanisms that cannot be provided in a higher layer.

<span class="h4"><a class="selflink" id="section-3.3.3" href="#section-3.3.3">3.3.3</a>.  Modifying RTP?</span>

   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?

   When RTP lives above UDP, modifying it to support congestion control
   might create some of the problems described in <a href="#section-3.1">Section 3.1</a>.  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 control.

   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, such as a reduced RTP header that removes any
   fields redundant with the new protocol's headers.

<span class="h4"><a class="selflink" id="section-3.3.4" href="#section-3.3.4">3.3.4</a>.  Designing a New Transport Protocol</span>

   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 as 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.



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<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Selling Congestion Control to Reluctant Applications</span>

   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 design requirements include a choice
   between different forms of congestion control, moderate overhead in
   the size of the packet header, and the use of Explicit Congestion
   Notification (ECN) and the ECN nonce [<a href="./rfc3540" title="&quot;Robust Explicit Congestion Notification (ECN) Signaling with Nonces&quot;">RFC3540</a>], which provide
   positive benefit to the application itself.

   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 [<a href="./rfc2309" title="&quot;Recommendations on Queue Management and Congestion Avoidance in the Internet&quot;">RFC2309</a>, <a href="#ref-FF99" title="&quot;Promoting the Use of End-to- End Congestion Control in the Internet&quot;">FF99</a>]
   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 toward 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.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Additional Design Considerations</span>

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

   o  Mobility: Mechanisms for multihoming 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 multihoming and
      mobility into the protocol itself.  The current version of DCCP
      [<a href="./rfc4340" title="&quot;Datagram Congestion Control Protocol (DCCP)&quot;">RFC4340</a>] includes no multihoming or mobility support.

   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 offer additional protection against DoS
      attacks.  Thus, in designing a new transport protocol, even one
      designed to provide minimal functionality, the requirements for



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      providing defense against DoS attacks and spoofing need to be
      considered.

   o  Interoperation with RTP: As <a href="#section-3.3.3">Section 3.3.3</a> describes, attention
      should be paid to any necessary or desirable changes in RTP when
      it is used over the new protocol, such as reduced RTP headers.

   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 time, and
      providing applications with some information about packet sequence
      numbers.

   o  Interactions with NATs and firewalls: NATs and firewalls don't
      interact well with UDP, with its lack of explicit flow setup and
      teardown and, in practice, the lack of well-known ports for many
      UDP applications.  Some of these issues are application specific;
      others should be addressed by the transport protocol itself.

   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
      [<a href="#ref-RUTS" title="&quot;http://www.ietf.org/proceedings/98dec/43rd-ietf- 98dec-142.html&quot;">RUTS</a>].  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 this document addresses the requirements relevant to our
      problem area that were brought up at the RUTS BOF.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Transport Requirements of Request/Response Applications</span>

   Up until now, this document has discussed the transport and
   congestion control requirements of applications that generate long-
   lived, large flows of unreliable datagrams.  This section discusses
   briefly the transport needs of another class of applications, those
   of request/response transfers where the response might be a small
   number of packets, with preferences that include both reliable
   delivery and a minimum of state maintained at the ends.  The reliable
   delivery could be accomplished, for example, by having the receiver
   re-query when one or more of the packets in the response is lost.
   This is a class of applications whose needs are not well-met by
   either UDP or by TCP.






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   Although there is a legitimate need for a transport protocol for such
   short-lived reliable flows of such request/response applications, we
   believe that the overlap with the requirements of DCCP is almost
   non-existent and that DCCP should not be designed to meet the needs
   of these request/response applications.  Areas of non-compatible
   requirements include the following:

   o  Reliability: DCCP applications don't need reliability (and long-
      lived applications that do require reliability are well-suited to
      TCP or SCTP).  In contrast, these short-lived request/response
      applications do require reliability (possibly client-driven
      reliability in the form of requesting missing segments of a
      response).

   o  Connection setup and teardown: Because DCCP is aimed at flows
      whose duration is often unknown in advance, it addresses
      interactions with NATs and firewalls by having explicit handshakes
      for setup and teardown.  In contrast, the short-lived
      request/response applications know the transfer length in advance,
      but cannot tolerate the additional delay of a handshake for flow
      setup.  Thus, mechanisms for interacting with NATs and firewalls
      are likely to be completely different for the two sets of
      applications.

   o  Congestion control mechanisms: The styles of congestion control
      mechanisms and negotiations of congestion control features are
      heavily dependent on the flow duration.  In addition, the
      preference of the request/response applications for a stateless
      server strongly impacts the congestion control choices.  Thus,
      DCCP and the short-lived request/response applications have rather
      different requirements both for congestion control mechanisms and
      for negotiation procedures.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Summary of Recommendations</span>

   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,
      explicit connection setup and teardown, reliable feature
      negotiation, and reliable congestion feedback.





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   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
      endpoints.

   o  Support for ECN should be included.  (Applications could still
      make the decision not to use ECN for a particular session.)

   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 resistant to spoofed connection
      attacks ("SYN floods").

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

   In 2002, there was judged to be a consensus about the need for a new
   unicast transport protocol for unreliable datagrams, and the next
   step was then the consideration of more detailed architectural
   issues.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Security Considerations</span>

   There are no security considerations for this document.  It does
   discuss a number of security issues in the course of problem
   analysis, such as DoS resistance and firewall traversal.  The
   security considerations for DCCP are discussed separately in
   [<a href="./rfc4340" title="&quot;Datagram Congestion Control Protocol (DCCP)&quot;">RFC4340</a>].

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  Acknowledgements</span>

   We would like to thank Spencer Dawkins, Jiten Goel, Jeff Hammond,
   Lars-Erik Jonsson, John Loughney, Michael Mealling, and Rik Wade for
   feedback on earlier versions of this document.  We would also like to
   thank members of the Transport Area Working Group and of the DCCP
   Working Group for discussions of these issues.









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Informative References

   [<a id="ref-BRS99">BRS99</a>]        Balakrishnan, H., Rahul, H., and S. Seshan, "An
                  Integrated Congestion Management Architecture for
                  Internet Hosts", SIGCOMM, Sept. 1999.

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

   [<a id="ref-PF01">PF01</a>]         Padhye, J. and S. Floyd, "Identifying the TCP Behavior
                  of Web Servers", SIGCOMM 2001.

   [<a id="ref-RFC2309">RFC2309</a>]      Braden, B., Clark, D., Crowcroft, J., Davie, B.,
                  Deering, S., Estrin, D., Floyd, S., Jacobson, V.,
                  Minshall, G., Partridge, C., Peterson, L.,
                  Ramakrishnan, K., Shenker, S., Wroclawski, J., and L.
                  Zhang, "Recommendations on Queue Management and
                  Congestion Avoidance in the Internet", <a href="./rfc2309">RFC 2309</a>, April
                  1998.

   [<a id="ref-RFC2326">RFC2326</a>]      Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
                  Streaming Protocol (RTSP)", <a href="./rfc2326">RFC 2326</a>, April 1998.

   [<a id="ref-RFC2525">RFC2525</a>]      Paxson, V., Allman, M., Dawson, S., Fenner, W.,
                  Griner, J., Heavens, I., Lahey, K., Semke, J., and B.
                  Volz, "Known TCP Implementation Problems", <a href="./rfc2525">RFC 2525</a>,
                  March 1999.

   [<a id="ref-RFC2914">RFC2914</a>]      Floyd, S., "Congestion Control Principles", <a href="https://www.rfc-editor.org/bcp/bcp41">BCP 41</a>,
                  <a href="./rfc2914">RFC 2914</a>, September 2000.

   [<a id="ref-RFC2960">RFC2960</a>]      Stewart, R., Xie, Q., Morneault, K., Sharp, C.,
                  Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M.,
                  Zhang, L., and V. Paxson, "Stream Control Transmission
                  Protocol", <a href="./rfc2960">RFC 2960</a>, October 2000.

   [<a id="ref-RFC3124">RFC3124</a>]      Balakrishnan, H. and S. Seshan, "The Congestion
                  Manager", <a href="./rfc3124">RFC 3124</a>, June 2001.

   [<a id="ref-RFC3168">RFC3168</a>]      Ramakrishnan, K., Floyd, S., and D. Black, "The
                  Addition of Explicit Congestion Notification (ECN) to
                  IP", <a href="./rfc3168">RFC 3168</a>, September 2001.

   [<a id="ref-RFC3261">RFC3261</a>]      Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                  Johnston, A., Peterson, J., Sparks, R., Handley, M.,
                  and E. Schooler, "SIP: Session Initiation Protocol",
                  <a href="./rfc3261">RFC 3261</a>, June 2002.



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   [<a id="ref-RFC3448">RFC3448</a>]      Handley, M., Floyd, S., Padhye, J., and J. Widmer,
                  "TCP Friendly Rate Control (TFRC): Protocol
                  Specification", <a href="./rfc3448">RFC 3448</a>, January 2003.

   [<a id="ref-RFC3540">RFC3540</a>]      Spring, N., Wetherall, D., and D. Ely, "Robust
                  Explicit Congestion Notification (ECN) Signaling with
                  Nonces", <a href="./rfc3540">RFC 3540</a>, June 2003.

   [<a id="ref-RFC3714">RFC3714</a>]      Floyd, S. and J. Kempf, "IAB Concerns Regarding
                  Congestion Control for Voice Traffic in the Internet",
                  <a href="./rfc3714">RFC 3714</a>, March 2004.

   [<a id="ref-RFC3758">RFC3758</a>]      Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
                  Conrad, "Stream Control Transmission Protocol (SCTP)
                  Partial Reliability Extension", <a href="./rfc3758">RFC 3758</a>, May 2004.

   [<a id="ref-RFC4340">RFC4340</a>]      Kohler, E., Handley, M., and S. Floyd, "Datagram
                  Congestion Control Protocol (DCCP)", <a href="./rfc4340">RFC 4340</a>, March
                  2006.

   [<a id="ref-RUTS">RUTS</a>]         Requirements for Unicast Transport/Sessions (RUTS)
                  BOF, Dec. 7, 1998.  URL
                  "<a href="http://www.ietf.org/proceedings/98dec/43rd-ietf-98dec-142.html">http://www.ietf.org/proceedings/98dec/43rd-ietf-</a>
                  <a href="http://www.ietf.org/proceedings/98dec/43rd-ietf-98dec-142.html">98dec-142.html</a>".



























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Authors' Addresses

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

   EMail: floyd@icir.org


   Mark Handley
   Department of Computer Science
   University College London
   Gower Street
   London WC1E 6BT
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   EMail: M.Handley@cs.ucl.ac.uk


   Eddie Kohler
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   UCLA Computer Science Department
   Los Angeles, CA 90095
   USA

   EMail: kohler@cs.ucla.edu






















<span class="grey">Floyd, et al.                Informational                     [Page 21]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-22" ></span>
<span class="grey"><a href="./rfc4336">RFC 4336</a>               Problem Statement for DCCP             March 2006</span>


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</pre>