Internet DRAFT - draft-kuehlewind-spud-use-cases

draft-kuehlewind-spud-use-cases







Network Working Group                                 M. Kuehlewind, Ed.
Internet-Draft                                          B. Trammell, Ed.
Intended status: Informational                                ETH Zurich
Expires: September 19, 2016                               March 18, 2016


      Use Cases for a Substrate Protocol for User Datagrams (SPUD)
                  draft-kuehlewind-spud-use-cases-01

Abstract

   This document identifies use cases for explicit cooperation between
   endpoints and middleboxes in the Internet under endpoint control.
   These use cases range from relatively low level applications
   (improving the ability for UDP-based protocols to traverse firewalls)
   through support for new transport services (in-flow prioritization
   for graceful in-network degradation of media streams).  They are
   intended to provide background for deriving the requirements for a
   Substrate Protocol for User Datagrams (SPUD), as discussed at the IAB
   Stack Evolution in a Middlebox Internet (SEMI) workshop in January
   2015 and the SPUD BoF session at IETF 92 in March 2015.

Status of This Memo

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   This Internet-Draft will expire on September 19, 2016.

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   Copyright (c) 2016 IETF Trust and the persons identified as the
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   This document is subject to BCP 78 and the IETF Trust's Legal
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   publication of this document.  Please review these documents



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   carefully, as they describe your rights and restrictions with respect
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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Principles and Assumptions  . . . . . . . . . . . . . . .   3
       1.1.1.  Trust and Integrity . . . . . . . . . . . . . . . . .   4
       1.1.2.  Endpoint Control  . . . . . . . . . . . . . . . . . .   4
       1.1.3.  Least Exposure  . . . . . . . . . . . . . . . . . . .   4
   2.  Firewall Traversal for UDP-Encapsulated Traffic . . . . . . .   4
     2.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .   5
     2.2.  Information Exposed . . . . . . . . . . . . . . . . . . .   5
     2.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .   6
     2.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .   7
     2.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .   7
   3.  On-Path State Lifetime Discovery and Management . . . . . . .   7
     3.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .   7
     3.2.  Information Exposed . . . . . . . . . . . . . . . . . . .   8
     3.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .   8
     3.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .   9
     3.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .   9
   4.  Path MTU Discovery  . . . . . . . . . . . . . . . . . . . . .  10
     4.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .  10
     4.2.  Information Exposed . . . . . . . . . . . . . . . . . . .  10
     4.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .  10
     4.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .  11
     4.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .  11
   5.  Low-Latency Service . . . . . . . . . . . . . . . . . . . . .  11
     5.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .  11
     5.2.  Information Exposed . . . . . . . . . . . . . . . . . . .  12
     5.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .  12
     5.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .  13
     5.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .  13
   6.  Reordering Sensitivity  . . . . . . . . . . . . . . . . . . .  13
     6.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .  14
     6.2.  Information Exposed . . . . . . . . . . . . . . . . . . .  14
     6.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .  15
     6.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .  15
     6.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .  15
   7.  Application-Limited Flows . . . . . . . . . . . . . . . . . .  15
     7.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .  15
     7.2.  Information Exposed . . . . . . . . . . . . . . . . . . .  16
     7.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .  16
     7.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .  17



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     7.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .  17
   8.  Priority Multiplexing . . . . . . . . . . . . . . . . . . . .  17
     8.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .  17
     8.2.  Information Exposed . . . . . . . . . . . . . . . . . . .  17
     8.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .  18
     8.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .  18
     8.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .  18
   9.  In-Band Measurement . . . . . . . . . . . . . . . . . . . . .  18
     9.1.  Problem Statement . . . . . . . . . . . . . . . . . . . .  18
     9.2.  Information Exposed . . . . . . . . . . . . . . . . . . .  19
     9.3.  Mechanism . . . . . . . . . . . . . . . . . . . . . . . .  20
     9.4.  Deployment Incentives . . . . . . . . . . . . . . . . . .  20
     9.5.  Security, Privacy, and Trust  . . . . . . . . . . . . . .  20
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  21
   11. Security Considerations . . . . . . . . . . . . . . . . . . .  21
   12. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  21
   13. Informative References  . . . . . . . . . . . . . . . . . . .  21
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  22

1.  Introduction

   This document describe use cases for a common Substrate Protocol for
   User Datagrams (SPUD) that could be used by superstrate transport or
   application protocols to explicitly expose information to and
   exchange information with middleboxes about application traffic and
   network conditions.

   For each use case, we first describe a problem that is difficult or
   impossible to solve with presently deployable protocols within the
   present Internet architecture.  We then discuss which information is
   exposed by endpoints about the traffic sent, and/or by SPUD-aware
   middleboxes and routers about the path that traffic will traverse.
   We also suggest potential mechanisms to use that exposed information
   at middleboxes and/or endpoints, in order to demonstrate the
   feasibility of using the exposed information to the given use
   case.The described mechanisms are not necessarily proposals for
   moving forward, nor do they necessarily represent the best approach
   for applying the exposed information, but should illustrate and
   motivate the applicability of the exposed information.  We further
   discuss incentives for deployment and any security, privacy, and
   trust issues that arise in exposing and/or making use of the
   information.

1.1.  Principles and Assumptions

   We make a few assumptions about first principles in elaborating these
   use cases




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1.1.1.  Trust and Integrity

   In this document, we assume no pre-existing trust relationship
   between the communication endpoints and any middlebox or router on
   the path.  We must therefore always assume that information that is
   exposed can be incorrect, and/or that the information will be
   ignored.

   This implies that while endpoints can verify the integrity of
   information exposed by remote endpoints, they cannot verify the
   integrity of information exposed by middleboxes.  Middleboxes cannot
   verify the integrity of any information at all.  In limited
   situations where a trust relationship can be established, e.g.,
   between a managed end-user device in an enterprise network and a
   corporate firewall, this verifiability can be improved.

1.1.2.  Endpoint Control

   We further assume that all information exposure by middleboxes
   happens under explicit endpoint control.  For that reason, the
   information exposed by middleboxes in this document takes only two
   forms.  In the first form, "accumulation", the endpoint creates space
   in the header for middleboxes to use to signal to the remote
   endpoint, which then sends the information back to the originating
   endpoint via a feedback channel.  In the second form, the middlebox
   sends a packed directly back to the endpoint with additional
   information about why a packet was dropped.  Other communications
   patterns may be possible, depending on the first principles chosen;
   this is a subject of future work.

1.1.3.  Least Exposure

   Additionally, this document follows the principle of least exposure:
   in each use case, we attempt to define the minimum amount of
   information exposed by endpoints and middleboxes required by the
   proposed mechanism to solve the identified problem.  In addition to
   being good engineering practice, this approach reduces the risk to
   privacy through inadvertent irrelevant metadata exposure, reduces the
   amount of information available for application fingerprinting, and
   reduces the risk that exposed information could otherwise be used for
   unintended purposes.

2.  Firewall Traversal for UDP-Encapsulated Traffic

   We presume, following an analysis of requirements in
   [I-D.trammell-spud-req], as well as trends in transport protocol
   development (e.g.  QUIC, the RTCWEB data channel) that UDP
   encapsulation will prove a viable approach for deploying new



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   protocols in the Internet.  This, however, leads us to a first
   problem that must be solved.

2.1.  Problem Statement

   UDP is often blocked by firewalls, or only enabled for a few well-
   known applications (e.g.  DNS, NTP).  Recent measurement work has
   shown that somewhere between 4% and 8% of Internet hosts may be
   affected by UDP impairment, depending on the population studied.
   Some networks (e.g. enterprise networks behind corporate firewalls)
   are far more likely to block UDP than others (e.g.  residential
   wireline access networks).

   In addition, some network operators assume that UDP is not often used
   for high-volume traffic, and is often a source of spoofing or
   reflected attack traffic, and is therefore safe to block or route-
   limit.  This assumption is becoming less true than it once was: the
   volume of (good) UDP traffic is growing, mostly due to voice and
   video (real-time) services (e.g.  RTCWEB) where TCP is not suitable.

   Even if firewall vendors and administrators are willing to change
   firewall rules to allow more diverse UDP services, it is hard to
   track session state for UDP traffic.  As UDP is unidirectional, it is
   unknown whether the receiver is willing to accept the connection.
   Further there is no way to figure how long state must be maintained
   once established.  To efficiently establish state along the path we
   need an explicit contract, as is done implicitly with TCP today.

2.2.  Information Exposed

   To maintain state in the network, it must be possible to easily
   assign each packet to a session that is passing a certain network
   node.  This state should be bound to something beyond the five-tuple
   to link packets together.  In [I-D.trammell-spud-req], we propose the
   use of identifiers for groups of packets, called ("tubes").  This
   allows for differential treatment of different packets within one
   five-tuple flow, presuming the application has control over
   segmentation and can provide requirements on a per-tube basis.  Tube
   IDs must be hard to guess: a tube ID in addition to a five-tuple as
   an identifier, given significant entropy in the tube ID, provides an
   additional assurance that only devices along the path or devices
   cooperating with devices along the path can send packets that will be
   recognized by middleboxes and endpoints as valid.

   Further, to maintain state, the sender must explicitly indicate the
   start and end of a tube to the path, while the receiver must confirm
   connection establishment.  This, together with the first packet
   following the confirmation, provides a guarantee of return



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   routability; i.e. that the sender is actually at the address it says
   it is.  This implies all SPUD tubes must be bidirectional, or at
   least support a feedback channel for this confirmation.  Even though
   UDP is not a bidirectional transport protocol, often services on top
   of UDP are bidirectional anyway.  Even if not, we only require one
   packet to acknowledge a new connection.  This is low overhead for
   this basic security feature.  This connection set-up should not
   impose any additional start-up latency, so the sender must be also
   able to send payload data in the first packet.

   If a firewall blocks a SPUD packet, it can be beneficial for the
   sender to know why the packet was blocked.  Therefore a SPUD-aware
   middlebox should be able to send error messages.  Such an error
   message can either be sent directly to the sender itself, or
   alternatively to the receiver that can decide to forward the error
   message to a sender or not.

2.3.  Mechanism

   A firewall or middlebox can use the tube ID as an identifier for its
   session state information.  If the tube ID is large enough it will be
   hard for a non- eavesdropping attacker to guess the ID.

   If a firewall receives a SPUD message that signals the start of a
   connection, it can decide to establish new state for this tube.
   Alternatively, it can also forward the packet to the receiver and
   wait if the connection is wanted before establishing state.  To not
   require forwarding of unknown payload, a firewall might want to
   forward the initial SPUD packet without payload and only send the
   full packet if the connection has be accepted by the receiver.

   The firewall must still maintain a timer to delete the state of a
   tube if no packets were received for a while.  However, if a end
   signal is received the firewall can remove the state information
   faster.

   If a firewall receives a SPUD message which does not indicate the
   start of a new tube and no state is available for this tube, it may
   decide to block the traffic.  This can happen if the state has
   already timed out or if the traffic was rerouted.  In addition a
   firewall may send an error message to the sender or the receiver
   indicating that no state information is available.  If the sender
   receives such a message it can resend a start signal (potentially
   together with other tube state information) and continue its
   transmission.






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2.4.  Deployment Incentives

   The ability to use existing firewall management best practices with
   new transport services over SPUD is necessary to ensure the
   deployability of SPUD.  In today's Internet, application developers
   really only have two choices for transport protocols: TCP, or
   transports implemented at the application layer and encapsulated over
   UDP.  SPUD provides a common shim layer for the second case, and the
   firewall traversal facility it provides makes these transports more
   likely to deploy.

   It is not expected that the information provided by SPUD will enable
   all generic UDP-encapsulated transports to safely pass firewalls.
   However, it does make state handling easier for new services that a
   firewall administrator is willing to allow.

2.5.  Security, Privacy, and Trust

   The tube ID is scoped to the five-tuple.  While this makes the tube
   ID useless for session mobility, it does mean that the valid ID space
   is sufficiently sparse to maintain the "hard to guess" property, and
   prevents tube IDs from being misused to track flows from the same
   endpoint across multiple addresses.  This limitation may need further
   discussion.

   By providing information about connection setup, SPUD exposes
   information equivalent to that available in the TCP header.  It makes
   connection lifetime information explicit and accessible without
   specific higher-layer/application- level knowledge.

3.  On-Path State Lifetime Discovery and Management

   Once the problem of connection setup is solved, the problem arises of
   managing the lifetime of state associated with that connection at
   various devices along the path: NAT and stateful firewall state
   timeouts are a common cause of connectivity issues in the Internet.

3.1.  Problem Statement

   Devices along the path that must keep state in order to function
   cannot assume that signals tearing down a connection are provided
   reliably.  This is also the case for current TCP traffic.  Therefore,
   all stateful on-path devices must implement a mechanism to remove the
   state if no traffic is seen for a given flow or tube for a while.
   Usually this is implemented by maintaining a timeout since the last
   observed packet.





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   If the timeouts are set too low, on-path state might be discarded
   while the endpoint connection is still alive; in the case of
   firewalls and NATs, this can lead to unreliable connectivity.  The
   common solution to this problem is for applications or transport
   protocols that do not have any productive traffic to send to send
   "heartbeat" or "keep-alive" packets to reset the state timeout along
   the path.  However, since the minimum timeout along the path is
   unknown to the endpoint, implementers of transport and application .
   A default value of 150ms is commonly used today.  This represents a
   fairly rapid generation of nonproductive traffic, and is especially
   onerous on battery- powered mobile devices, which must wake up radios
   and switch to a higher-power mode to transmit these nonproductive
   packets, leading to suboptimal power usage and shorter battery life.

3.2.  Information Exposed

   SPUD can be used to request that SPUD-aware middleboxes along the
   path expose their minimum state timeout value.  Here, the sending
   endpoint sends a "accumulate minimum timeout" request along with some
   scratch space for middleboxes to place their timeout information in.
   Each middlebox inspects this value, and writes its own timeout only
   if lower than the present value.

   Applications may also send a "timeout proposal" to devices along the
   path using a SPUD declaration that a given tube will send a packet at
   least once per interval, and if no packet is seen within that
   interval, it is safe to tear down state.

   These two declarations may be used together, with middleboxes willing
   to use the application's value setting their timeouts on a per-tube
   basis, or exposing a lower timeout value to allow the application to
   adjust.

3.3.  Mechanism

   If a SPUD-aware middlebox that uses a timeout to clean up per-tube
   state receives a SPUD minimum timeout accumulation, it should expose
   its own timeout value if smaller than the one already given.
   Alternatively, if a value is already given, it might decide to use
   the given value as timeout for the state information of this tube.
   An endpoint receiving an accumulated minimum timeout should send it
   back to its remote peer via a feedback channel.  Timeouts on each
   direction of a connection between two endpoints may, of course, be
   different, and are handled separately by this mechanism.

   If a SPUD-aware middlebox that uses a timeout to clean up per-tube
   state receives a timeout proposal, it should set its timeout
   accordingly, subject to its own policy and configuration.



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   These mechanisms are of course completely advisory: there may be non-
   SPUD aware middleboxes on path which will ignore any proposed timeout
   and not expose their timeout information, and middleboxes must be
   configured with maximum timeout proposal they will accept in order to
   defend against state exhaustion attacks.

   Endpoints must therefore be combine the use of these signals endpoint
   with a dynamic timeout discovery and adaptation mechanism, which uses
   the signals to set initial guesses as to the path timeout.

3.4.  Deployment Incentives

   Initially, if not widely deployed, there will be not much benefit to
   using this extension.

   However, we can assume that there are usually only a small number of
   middleboxes on a given network path that hold per-tube state
   information.  Endpoints have an incentive to request minimum timeout
   and to propose timeouts to improve convergence time for dynamic
   timeout adaptation mechanisms, and middleboxes have an incentive to
   cooperate to improve reliability of connections as well as state
   management.  It is therefore likely that if information is exposed by
   a middlebox, this information is correct and can be used.

   The more SPUD gets deployed, the more often endpoints will be able to
   set the heartbeat interval correctly.  This will reduce the amount of
   unproductive traffic as well as the number of reconnections that
   cause additional latency.

   Likewise, SPUD-aware middleboxes that expose timeout information are
   able to handle timeouts more flexibly, e.g. announcing lower timeout
   values when they have less space available for new state.  Further if
   an endpoint announces a low pre-set value because the endpoint knows
   that it will only have short idle periods, the timeout interval could
   be reduced.

3.5.  Security, Privacy, and Trust

   Timeout proposals increase the risk of state exhaustion attacks for
   SPUD-aware middleboxes that naively follow them.  Likewise,
   accumulated minimum timeouts could be used by malicious middleboxes
   to induce floods of useless heartbeat traffic along the path, and/or
   exhaust resources on endpoints that naively follow them.  All timeout
   proposals and minimum timeouts must therefore be inputs to a dynamic
   timeout selection process, both at endpoints and on-path devices,
   which use these signals as hints but clamp their timeouts to sane
   values set by local policy.




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   While device timeout and heartbeat interval are generally not linked
   to privacy-sensitive information, a timeout proposal may add a number
   of bits of entropy to an endpoint's unique fingerprint.  It is
   therefore advisable to suggest a small number of useful timeout
   proposals, in order to reduce this value's contribution to an
   endpoint fingerprint.

4.  Path MTU Discovery

   Similar to the state timeout problem is the Path MTU problem:
   differing MTUs on different devices along the path can lead to
   fragmentation or connectivity issues.  This problem is made worse by
   the increasing proliferation of tunnels in the Internet, which reduce
   the MTU by the amount required for tunnel headers.

4.1.  Problem Statement

   In order to efficiently send packets along a path end to end, they
   must be sized to fit in the MTU of the "narrowest" link along the
   path.  Algorithms for path MTU discovery have been defined and
   standardized for a quarter century, in [RFC1191] for IPv4 and
   [RFC1981] for IPv6, but they are not often implemented due in part to
   widespread impairment of ICMP.  Packetization Layer Path MTU
   Discovery [RFC4821] (PLPMTUD) is a more recent attempt to solve the
   problem, which has the advantage of being transport-protocol
   independent and functional without ICMP feedback.  SPUD, as a shim
   between UDP and superstrate transport protocols, is at the right
   place in the stack to implement PLPMTUD, and explicit cooperation can
   enhance its operation.

4.2.  Information Exposed

   SPUD can be used to request that SPUD-aware middleboxes along the
   path expose their next-hop path MTU value.  Here, the sending
   endpoint sends a "accumulate minimum MTU" request along with some
   scratch space for middleboxes to place the next-hop MTU for the given
   tube.  Each middlebox inspects this value, and writes its own next-
   hop MTU only if lower than the present value.

   A SPUD-aware middlebox that receives a packet that is too big for the
   next-hop MTU can send back a signal associated with the tube directly
   to the sender, including the next-hop MTU.

4.3.  Mechanism

   PLPMTUD functions by dynamically increasing the size of packets sent,
   and reacting to the loss of the first "too large" packet as an MTU
   reduction signal, instead of a congestion signal.  This must be



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   implemented in cooperation with the superstrate transport protocol,
   as it is responsible for how non-MTU-related loss is treated.

   When an endpoint receives an accumulated minimum MTU, it should
   should send it back to its remote peer via a feedback channel.  The
   minimum of this value and any direct next-hop MTU signals received
   from SPUD-aware middleboxes can be used as a hint to the sender's
   PLPMTUD process, as a likely upper bound for path MTU associated with
   a tube.

4.4.  Deployment Incentives

   As with state lifetime discovery, these signals are of little initial
   utility to endpoints before SPUD-aware middleboxes are deployed.
   However, SPUD-aware middleboxes that sit at potential MTU breakpoints
   along a path, either those which terminate tunnels or bridge networks
   with two different link types, have an incentive to improve
   reliability by responding to accumulation requests and sending next-
   hop MTU messages to SPUD-aware endpoints.

4.5.  Security, Privacy, and Trust

   As with state lifetime discovery, Minimum MTU and next-hop MTU
   signals could be used by malicious middleboxes to set the endpoint's
   maximum packet size to inefficiently small sizes, if the endpoint
   follows them naively.  For that reason, endpoints should use this
   information only as hints to improve the operation of PLPMTUD, and
   may probe above the value derived from the SPUD- supplied information
   when deemed appropriate by endpoint policy or transport protocol
   requirements.

5.  Low-Latency Service

5.1.  Problem Statement

   Networks are often optimized for low loss rates and high throughput
   by providing large buffers that can absorb traffic spikes and rate
   variations while holding enough data to keep the link full.  This is
   beneficial for applications like high-priority bulk transfer, where
   only the total transfer time is of interest.  High-volume interactive
   applications, such as videoconferencing, however, have very different
   requirements.  Usually these applications can tolerate higher loss
   rates, while having hard latency requirements.

   Large network buffers may induce high queuing delays due to cross
   traffic using loss-based congestion control, which must periodically
   fill the buffer to induce loss during probing for additional
   bandwidth.  This queueing delay can negatively impact the quality of



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   experience for competing interactive applications, even making them
   unusable.

5.2.  Information Exposed

   The simplest mechanism for solving this problem is to separate loss-
   sensitive from latency-sensitive traffic, as proposed using DSCP
   codepoints in [I-D.you-tsvwg-latency-loss-tradeoff].  This signal
   could also be emitted as a per-packet signal within SPUD, since DSCP
   codepoints are often used for internal traffic engineering and
   therefore cleared at network borders.  This indication does not
   prioritize one kind of traffic over the other: while loss- sensitive
   traffic might face larger buffer delay but lower loss rate, latency-
   sensitive traffic has to make exactly the opposite tradeoff.

   An endpoint can also indicate a maximum acceptable single-hop
   queueing delay per tube, expressed in milliseconds.  While this
   mechanism does not guarantee that sent packets will experience less
   than the requested delay due to queueing delay, it can significantly
   reduce the amount of traffic uselessly sitting in queues, since at
   any given instance only a small number of queues along a path
   (usually only zero or one) will be full.

5.3.  Mechanism

   A middlebox may use the loss-/latency tradeoff signal to assign
   packet to the appropriate type of service, if different services are
   implemented at this middlebox.  Traffic not indicating a low loss or
   low latency preference would still be assigned to today's best-effort
   service, while a new low latency service would be introduced in
   addition.

   The simplest implementation of such a low latency service (without
   disturbing existing traffic) is to manage traffic with the latency-
   sensitive flag set in a separate queue.  This queue either, in
   itself, provides only a short buffer which induces a hard limit for
   the maximum (per-queue) delay or uses an AQM (such as PIE/CoDel) that
   is configured to keep the queuing delay low.

   In such a two-queue system the network provider must decide about
   bandwidth sharing between both services, and might or might not
   expose this information.  Initially there will only be a few flows
   that indicate low latency preference.  Therefore at the beginning
   this service might have a low maximum bandwidth share assigned in the
   scheduler.  However, the sharing ratio should be adapted to the
   traffic load/number of flows in each service class over long
   timescales.




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   Applications and endpoints setting the latency sensitivity flag on a
   tube must be prepared to experience relatively higher loss rates on
   that tube, and should use techniques such as Forward Error Correction
   (FEC) to cope with these losses.

   If a maximum per-hop delay is indicated by the sender, a SPUD- aware
   router might drop any packet which would be placed in a queue that
   has more than the maximum single-hop delay at that point in time
   before queue admission.  Thereby the overall congestion can be
   reduced early instead of withdrawing the packet at the receiver after
   it has blocked network resources for other traffic.

   A transport protocol at an endpoint indicating the maximum per-hop
   delay must be aware that is might face higher loss rates under
   congestion than competing traffic on the same bottleneck.

5.4.  Deployment Incentives

   Application developers go to a great deal of effort to make latency-
   sensitive traffic work over today's Internet.  However, if large
   delays are induced by the network, an application at the endpoint
   cannot do much.  Therefore applications can benefit from further
   support by the network.

   Network operators have already realized a need to better support low
   latency services.  However, they want to avoid any service
   degradation for existing traffic as well as risking stability due to
   large configuration changes.  Introducing an additional service for
   latency-sensitive traffic that can exist in parallel to today's
   network service helps this problem.

5.5.  Security, Privacy, and Trust

   An application cannot benefit from wrongly indicating loss- or
   latency- sensitivity, as it has to make a tradeoff between low loss
   and potential high delay or low delay and potential high loss.

   A simple classification of traffic as loss- or latency-sensitive does
   not expose privacy-critical information about the user's behavior;
   indeed, it exposes far less than presently used by DPI-based traffic
   classifiers that would be used to determine the latency sensitivity
   of traffic passing a middlebox.

6.  Reordering Sensitivity







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6.1.  Problem Statement

   TCP's fast retransmit mechanism interprets the reception of three
   duplicated acknowledgement (where the acknowledgement number is the
   same than in the previous acknowledgement) as a signal for loss
   detection.  However, a missing packet in the sequence number space
   must not always be lost.  Simple reordering where one packet takes a
   longer path than (at least three) subsequent packets can have the
   same effect.

   In addition in TCP, loss is an implicit signal for network
   congestion.  Therefore the reception of three duplicated
   acknowledgement will cause a TCP sender to reduce its sending rate.
   To avoid unnecessary performance decreases, today's in-network
   mechanisms usually aim to avoid reordering.  However, this
   complicates these mechanism significantly and usually requires per-
   flow state, e.g. in case of Equal Cost Multipath (ECMP) routing where
   a hash of the 5 tuple would need to be mapped to the right path.

   Even though the majority of traffic in the Internet is still TCP, it
   is likely that new protocols will be design such that they are (more)
   robust to reordering.  Further with an increasing deployment of ECN,
   even TCP's congestion control reaction based on duplicated
   acknowledgements could be relaxed (e.g. by reducing the sending rate
   gradually depending on the number of lost packets).

   However, as middlebox can not know if a certain traffic flow is
   sensitive to reordering or not, they have to treat all traffic as
   equally and try to always avoid reordering.  (This does not only
   complicate these mechanism but might also block the deployment of new
   services.)

6.2.  Information Exposed

   Reordering-sensitivity is a per tube signal (as reordering can only
   happen with a flow multiple packets).  However, to avoid state in
   middlebox, it would be beneficial to have a reordering-sensitive flag
   in each packet.

   A transport should set the bit if it is not sensitive to reordering,
   e.g. if it uses a more advance mechanism (than duplicated
   acknowledgement) for loss detection, or if the congestion control
   reaction to this signal imposes only a small performances penalty, or
   if the flow is short enough that it will not impact its performance.







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6.3.  Mechanism

   A middlebox that implement an in-network function that could lead to
   varying end-to-end delay and reordering (as packets might overtake
   each other on different paths or within the network device), do not
   need to perform any additional action if the reordering-sensitivity
   flag is not set.  However, if the flag is set, the middlebox should
   avoid reordering by e.g. holding per- tube state and make sure that
   all packets belonging to the same tube will not be re-ordered.

6.4.  Deployment Incentives

   Today by default middlebox assume that all traffic is reordering-
   sensitive which complicates certain in-network mechanism or might
   also block the deployment of new services.  If a middlebox would know
   that certain traffic is not reordering-sensitive, it could reduce
   state, speed-up processing, or even implement new services.

   Applications that are not loss-sensitive (because they e.g. uses FEC)
   usually are also not reordering-sensitive.  At the same time these
   application are often sensitive to latency.  If the transport handles
   reordering appropriately and signal this semantic information to the
   network, the appropriate network treatment can likely also result in
   lower end-to-end or at least enables the network device to impose any
   additional delay (e.g. to set up state) on these packets.

6.5.  Security, Privacy, and Trust

   No trust relationship is needed as the provided information do not
   results in a preferential treatment.  Only transport semantics are
   exposed that to not contain any private information.

7.  Application-Limited Flows

7.1.  Problem Statement

   Many flows are application-limited, where the application itself
   adapts the limit to changing traffic conditions or link
   characteristics, such as with unicast adaptive bitrate streaming
   video.  This adaptation is difficult, since TCP cross-traffic will
   often probe for available bandwidth more aggressively than the
   application's control loop.  Further complicating the situation is
   the fact that rate adaptation may have negative effects on the user's
   quality of experience, and should therefore be done infrequently.







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7.2.  Information Exposed

   A SPUD endpoint sending application-limited traffic can provide an
   explicit per-tube indication of the maximum intended data rate needed
   by the current encoding or data source.  If the bottleneck device is
   SPUD-aware, it can use this information to decide how to correctly
   treat the tube, e.g. setting a rate limit or scheduling weight if
   served from its own queue.

   A SPUD endpoint could also send a "minimum rate limit accumulation"
   request, similar to the other accumulation requests outlined above,
   where SPUD-aware routers and middleboxes could note the maximum
   bandwidth available to a tube.  Receiving this signal on a feedback
   channel could allow a sender to more quickly adapt its sending rate.
   This rate limit information might be derived from local per-flow or
   per-tube rate limit policy, as well as from current information about
   load at the router.

   These signals can be sent throughout the lifetime of the flow, to
   help adapt to changing application demands and/or network conditions.

7.3.  Mechanism

   Maximum expected data rate exposed by the endpoints could be used to
   make routing decisions and queue selection decisions at SPUD-aware
   routers, if different paths or queues with different capacity, delay,
   and load characteristics are available.

   A SPUD-aware router that indicates a rate limit can be used by the
   sender to choose an encoding.  However, the sender should still
   implement a mechanism to probe for available bandwidth to verify the
   provided information.  As a certain rate limit is expected, the
   sender should probe carefully around this rate.

   These mechanisms can also be used for rate increases.  If a sender
   receives an indication that more bandwidth is available it should
   probe carefully, instead of switching to the higher rate immediately,
   and decrease its sensitivity to loss (e.g. through the use of
   additional FEC) which will provide additional protection as soon as
   the new capacity limit is reached.  Likewise, a SPUD- aware router
   that receives an indication that a flow intends to increase its might
   prioritize this flow for a certain (short) time to enable a smoother
   transition.








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7.4.  Deployment Incentives

   Endpoints that indicate maximum sending rate for application-limited
   traffic on SPUD-aware networks allow the operators of those networks
   to better handle traffic.  This can benefit the service quality and
   increase the user's satisfaction with the provided network service.

   Currently applications have no good indication when to change their
   coding rate.  Rate increases are especially hard.  Further, frequent
   rate changes should be avoided for quality of experience.
   Cooperative indication of intended and available sending rate for
   application-limited flows can simplify probing, and provide signals
   beyond loss to react effectively to congestion.

7.5.  Security, Privacy, and Trust

   Both endpoints and SPUD-aware middleboxes should react defensively to
   rate limit and rate intention information.  Endpoints and middleboxes
   should use measurement and probing to verify that rate information is
   accurate, but the exposed rate information can be used as hints to
   routing, scheduling, and rate determination processes.

8.  Priority Multiplexing

8.1.  Problem Statement

   Many services require multiple parallel transmissions to transfer
   different kinds of data which have clear priority relationships among
   them.  For example, in WebRTC, audio frames should be prioritized
   over video frames.  Sometimes these transmissions happen in different
   flows, and sometimes some packets within a flow have higher priority
   than others, for example I-frames in video transmissions.  However,
   current networks will treat all packets the same in case of
   congestion and might e.g. drop audio packets while video and control
   traffic are still transmitted.

8.2.  Information Exposed

   A SPUD sender may indicate a that one tube should "yield" to another,
   i.e.  that it should have lower relative priority than another tube
   in the same flow.  Similarly, individual packets within a tube could
   be marked as having lower priority.  This information can be used to
   preferentially drop less important packets e.g. carrying information
   that could be recovered by FEC.

   With a stronger integration of codec and transport protocols, SPUD
   could even indicate more fine-grained priority levels to provide
   automatic graceful degradation of service within the network itself.



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8.3.  Mechanism

   Designing a general-purpose mechanism that maps relative priorities
   from the yield information exposed via SPUD to correct per-tube and
   per-packet treatment at any point in the Internet, is an extremely
   hard problem and a possible subject for future research.  It appears
   impossible at this writing to design a straightforward mapping
   function from these relative priorities per- flow to absolute
   priorities across flows in a fair way.

   However, in the not-uncommon case that exists in many access
   networks, where the bottleneck link has per-user queues and can
   enforce per-user fairness, the relative priorities can be mapped to
   absolute priorities, and simple priority queueing at the bottleneck
   can be used.  Lower priority packets within a tube, however, should
   be assigned to the tube's priority class, and preferentially dropped
   instead, e.g. using a different drop threshold at the queue.

8.4.  Deployment Incentives

   Deployment incentives for priority multiplexing are similar to those
   for bandwidth declaration for app-limited flows as in Section 7.4:
   endpoints that correctly declare priority information will experience
   better quality of service on SPUD-enabled networks, and SPUD-enabled
   networks get information that allows them to better manage traffic.

8.5.  Security, Privacy, and Trust

   Since yield information can only be used to disadvantage an
   application's traffic relative to its own traffic, there is no
   incentive for applications to declare incorrect yielding.

   The pattern and relative volume of traffic in different yield classes
   may be used to "fingerprint" certain applications, though it is not
   clear whether this provides additional information beyond that
   contained inter-packet delay and volume patterns.

9.  In-Band Measurement

9.1.  Problem Statement

   The current Internet protocol stack has very limited facilities for
   network measurement and diagnostics.  The only explicit measurement
   feature built into the stack is ICMP Echo ("ping").  In the meantime,
   the Internet measurement community has defined many inference- and
   assumption-based approaches for getting better information out of the
   network: traceroute and BGP looking glasses for topology information,
   TCP sequence number and TCP timestamp based approaches for latency



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   and loss estimation, and so on.  Each of these uses values placed on
   the wire for the internal use of the protocol, not for measurement
   purposes, and do not necessarily apply to the deployment of new
   protocols or changes to the use of those values by protocol
   implementations.  Approaches involving the encryption of transport
   protocol and application headers (indeed, including that the authors
   advance in [I-D.trammell-spud-req]) will break most of these, as
   well.

   Replacing the information used for measurement with values defined
   explicitly to be used for measurement in a transport protocol
   independent way allows explicit endpoint control of measurability and
   measurement overhead.

   We note that current work in IPPM [I-D.ietf-ippm-6man-pdm-option]
   proposes a roughly equivalent, IPv6-only, kernel-implementation-only
   facility.

9.2.  Information Exposed

   The "big five" metrics - latency, loss, jitter, data rate / goodput,
   and reordering - can be measured using a relatively simple set of
   primitives.  Packet receipt acknowledgment using a cumulative nonce
   echo allows both endpoint and on-path measurement of loss and
   reordering as well as goodput (when combined with layer 3 packet
   length headers).  A timestamp echo facility, analogous to TCP's
   timestamp option but using an explicitly defined, constant-rate clock
   and exposure of local delta (time between receipt and subsequent
   transmission).

   The cumulative nonce echo consists of two values: a number
   identifying a given packet (nonce), which also identifies all
   retransmissions of the packet, and a number which is the sum of all
   packet identifiers received from the remote endpoint (echo), modulo
   the maximum value of the echo field.  Nonces need not be sequential,
   or even monotonic, but two packets with the same nonce should not be
   simultaneously in flight.  These are exposed on a per-packet basis,
   but need not appear on every packet in the tube or flow, with the
   caveat that lower sampling rates lead to lower sensitivity.

   The timestamp echo consists of three values: The time in terms of
   ticks of a constant rate clock that a packet is sent, the echo of the
   last such timestamp received from the remote endpoint, and the number
   of ticks of the sender's clock between the receipt of the last
   timestamp from the remote endpoint and the transmission of the packet
   containing the echo.  This last delta value is the missing link in
   TCP sequence number based and timestamp option based latency
   estimation.



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   The information exposed is roughly equivalent than that currently
   exposed by TCP as a side effect of its operation, but defined such
   that they are explicitly useful for measurement, useful regardless of
   transport protocol, and such that information exposure is in the
   explicit control of the endpoint (when the superstrate transport
   protocol's headers are encrypted).

9.3.  Mechanism

   The nonce and timestamp echo information, emitted as per-packet
   signals in the SPUD header, can be used by any device which can see
   it to estimate performance metrics on a per-tube basis.  This
   includes both remote endpoints, as well as passive performance
   measurement devices colocated with network gateways.

9.4.  Deployment Incentives

   Initial deployment of this facility is most likely in closed networks
   such as enterprise data centers, where a single administrative entity
   owns the network and the endpoints, can control which flows and tubes
   are annotated with measurement information, and can benefit from the
   additional insight given during network troubleshooting by explicit
   measurement headers.

   Further, since the provided measurement information is exposed by
   SPUD to the far-endpoint, it can be used for performance enhancement
   on these layers.  Once the facility is deployed in SPUD-aware
   endpoints, it can also be used for inter-network and cross-Internet
   performance measurement and debugging (replacing today's processing-
   intensive DPI mechanisms).

9.5.  Security, Privacy, and Trust

   The cumulative nonce and timestamp echo leaks no more information
   about the traffic than the TCP header does.  Indeed, since the
   cumulative nonce does not include sequence number information or
   other protocol-internal information, it allows passive measurement of
   loss and latency without giving measurement devices access to
   information they could use to spoof valid packets within a transport
   layer connection.

   In order to prevent middleboxes from modifying measurement-relevant
   information, these per-packet signals will need to be integrity
   protected by SPUD.

   Performance measurement boxes at gateways which observe and aggregate
   these signals will necessarily need to trust their accuracy, but can




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   verify their plausibility by calculating nonce sums and synchronizing
   timing clocks.

10.  IANA Considerations

   This document has no actions for IANA.

11.  Security Considerations

   Security and privacy considerations for each use case are given in
   the corresponding Security, Privacy, and Trust subsection.

12.  Acknowledgments

   This document grew in part out of discussions of initial use cases
   for middlebox cooperation at the IAB SEMI Workshop and the IETF 92
   SPUD BoF; thanks to the participants.  Some use case details came out
   of discussions with the authors of the [I-D.trammell-spud-req]: in
   addition to the editors of this document, David Black, Ken Calvert,
   Ted Hardie, Joe Hildebrand, Jana Iyengar, and Eric Rescorla.
   Section 9 is based in part on discussions and ongoing work with Mark
   Allman and Rob Beverly.

   This work is supported by the European Commission under Horizon 2020
   grant agreement no. 688421 Measurement and Architecture for a
   Middleboxed Internet (MAMI), and by the Swiss State Secretariat for
   Education, Research, and Innovation under contract no. 15.0268.  This
   support does not imply endorsement.

13.  Informative References

   [I-D.hildebrand-spud-prototype]
              Hildebrand, J. and B. Trammell, "Substrate Protocol for
              User Datagrams (SPUD) Prototype", draft-hildebrand-spud-
              prototype-03 (work in progress), March 2015.

   [I-D.ietf-ippm-6man-pdm-option]
              Elkins, N. and M. Ackermann, "IPv6 Performance and
              Diagnostic Metrics (PDM) Destination Option", draft-ietf-
              ippm-6man-pdm-option-01 (work in progress), October 2015.

   [I-D.trammell-spud-req]
              Trammell, B. and M. Kuehlewind, "Requirements for the
              design of a Substrate Protocol for User Datagrams (SPUD)",
              draft-trammell-spud-req-02 (work in progress), March 2016.






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   [I-D.you-tsvwg-latency-loss-tradeoff]
              You, J., Welzl, M., Trammell, B., Kuehlewind, M., and K.
              Smith, "Latency Loss Tradeoff PHB Group", draft-you-tsvwg-
              latency-loss-tradeoff-00 (work in progress), March 2016.

   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
              DOI 10.17487/RFC1191, November 1990,
              <http://www.rfc-editor.org/info/rfc1191>.

   [RFC1981]  McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
              for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August
              1996, <http://www.rfc-editor.org/info/rfc1981>.

   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
              <http://www.rfc-editor.org/info/rfc4821>.

Authors' Addresses

   Mirja Kuehlewind (editor)
   ETH Zurich
   Gloriastrasse 35
   8092 Zurich
   Switzerland

   Email: mirja.kuehlewind@tik.ee.ethz.ch


   Brian Trammell (editor)
   ETH Zurich
   Gloriastrasse 35
   8092 Zurich
   Switzerland

   Email: ietf@trammell.ch
















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