Transport Services (tsv) B. Briscoe, Ed.
Internet-Draft Simula Research Lab
Intended status: Informational K. De Schepper
Expires: January 9, 2017 Nokia Bell Labs
M. Bagnulo Braun
Universidad Carlos III de Madrid
July 8, 2016
Low Latency, Low Loss, Scalable Throughput (L4S) Internet Service:
Problem Statement
draft-briscoe-tsvwg-aqm-tcpm-rmcat-l4s-problem-02
Abstract
This document motivates a new service that the Internet could provide
to eventually replace best efforts for all traffic: Low Latency, Low
Loss, Scalable throughput (L4S). It is becoming common for _all_ (or
most) applications being run by a user at any one time to require low
latency. However, the only solution the IETF can offer for ultra-low
queuing delay is Diffserv, which only favours a minority of packets
at the expense of others. In extensive testing the new L4S service
keeps average queuing delay under a millisecond for _all_
applications even under very heavy load, without sacrificing
utilization; and it keeps congestion loss to zero. It is becoming
widely recognized that adding more access capacity gives diminishing
returns, because latency is becoming the critical problem. Even with
a high capacity broadband access, the reduced latency of L4S
remarkably and consistently improves performance under load for
applications such as interactive video, conversational video, voice,
Web, gaming, instant messaging, remote desktop and cloud-based apps
(even when all being used at once over the same access link). The
insight is that the root cause of queuing delay is in TCP, not in the
queue. By fixing the sending TCP (and other transports) queuing
latency becomes so much better than today that operators will want to
deploy the network part of L4S to enable new products and services.
Further, the network part is simple to deploy - incrementally with
zero-config. Both parts, sender and network, ensure coexistence with
other legacy traffic. At the same time L4S solves the long-
recognized problem with the future scalability of TCP throughput.
This document explains the underlying problems that have been
preventing the Internet from enjoying such performance improvements.
It then outlines the parts necessary for a solution and the steps
that will be needed to standardize them. It points out opportunities
that will open up, and sets out some likely use-cases, including
ultra-low latency interaction with cloud processing over the public
Internet.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. The Application Performance Problem . . . . . . . . . . . 3
1.2. The Technology Problem . . . . . . . . . . . . . . . . . 4
1.3. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5
1.4. The Standardization Problem . . . . . . . . . . . . . . . 7
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 9
2.1. Why These Primary Components? . . . . . . . . . . . . . . 9
2.2. Why Not Alternative Approaches? . . . . . . . . . . . . . 10
3. Opportunities . . . . . . . . . . . . . . . . . . . . . . . . 11
3.1. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . 12
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
5. Security Considerations . . . . . . . . . . . . . . . . . . . 14
5.1. Traffic (Non-)Policing . . . . . . . . . . . . . . . . . 14
5.2. 'Latency Friendliness' . . . . . . . . . . . . . . . . . 14
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5.3. ECN Integrity . . . . . . . . . . . . . . . . . . . . . . 15
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16
7. References . . . . . . . . . . . . . . . . . . . . . . . . . 16
7.1. Normative References . . . . . . . . . . . . . . . . . . 16
7.2. Informative References . . . . . . . . . . . . . . . . . 16
Appendix A. Required features for scalable transport protocols
to be safely deployable in the Internet (a.k.a. TCP
Prague requirements) . . . . . . . . . . . . . . . . 19
Appendix B. Standardization items . . . . . . . . . . . . . . . 23
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 26
1. Introduction
1.1. The Application Performance Problem
It is increasingly common for _all_ of a user's applications at any
one time to require low delay: interactive Web, Web services, voice,
conversational video, interactive video, instant messaging, online
gaming, remote desktop and cloud-based applications. In the last
decade or so, much has been done to reduce propagation delay by
placing caches or servers closer to users. However, queuing remains
a major, albeit intermittent, component of latency. When present it
typically doubles the path delay from that due to the base speed-of-
light. Low loss is also important because, for interactive
applications, losses translate into even longer retransmission
delays.
It has been demonstrated that, once access network bit rates reach
levels now common in the developed world, increasing capacity offers
diminishing returns if latency (delay) is not addressed.
Differentiated services (Diffserv) offers Expedited Forwarding
[RFC3246] for some packets at the expense of others, but this is not
applicable when all (or most) of a user's applications require low
latency.
Therefore, the goal is an Internet service with ultra-Low queueing
Latency, ultra-Low Loss and Scalable throughput (L4S) - for _all_
traffic. Having motivated the goal of 'L4S for all', this document
enumerates the problems that have to be overcome to reach it.
It must be said that queuing delay only degrades performance
infrequently [Hohlfeld14]. It only occurs when a large enough
capacity-seeking (e.g. TCP) flow is running alongside the user's
traffic in the bottleneck link, which is typically in the access
network. Or when the low latency application is itself a large
capacity-seeking flow (e.g. interactive video). At these times, the
performance improvement must be so remarkable that network operators
will be motivated to deploy it.
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1.2. The Technology Problem
Active Queue Management (AQM) is part of the solution to queuing
under load. AQM improves performance for all traffic, but there is a
limit to how much queuing delay can be reduced by solely changing the
network; without addressing the root of the problem.
The root of the problem is the presence of standard TCP congestion
control (Reno [RFC5681]) or compatible variants (e.g. TCP Cubic
[I-D.ietf-tcpm-cubic]). We shall call this family of congestion
controls 'Classic' TCP. It has been demonstrated that if the sending
host replaces Classic TCP with a 'Scalable' alternative, when a
suitable AQM is deployed in the network the performance under load of
all the above interactive applications can be stunningly improved.
For instance, queuing delay under heavy load with the example DCTCP/
DualQ solution cited below is roughly 1 millisecond (1 ms) at the
99th percentile without losing link utilization. This compares with
5 to 20 ms on _average_ with a Classic TCP and current state-of-the-
art AQMs such as fq_CoDel [I-D.ietf-aqm-fq-codel] or
PIE [I-D.ietf-aqm-pie]. Also, with a Classic TCP, 5 ms of queuing is
usually only possible by losing some utilization.
It has been convincingly demonstrated [DCttH15] that it is possible
to deploy such an L4S service alongside the existing best efforts
service so that all of a user's applications can shift to it when
their stack is updated. Access networks are typically designed with
one link as the bottleneck for each site (which might be a home,
small enterprise or mobile device), so deployment at a single node
should give nearly all the benefit. Although the main incremental
deployment problem has been solved, and the remaining work seems
straightforward, there may need to be changes in approach during the
process of engineering a complete solution.
There are three main parts to the L4S approach (illustrated in
Figure 1):
2) Network: The L4S service needs to be isolated from the queuing
latency of the Classic service. However, the two should be able
to freely share a common pool of capacity. This is because there
is no way to predict how many flows at any one time might use each
service and capacity in access networks is too scarce to partition
into two. So a 'semi-permeable' membrane is needed that
partitions latency but not bandwidth. The Dual Queue Coupled AQM
[I-D.briscoe-aqm-dualq-coupled] is an example of such a semi-
permeable membrane.
Per-flow queuing such as in [I-D.ietf-aqm-fq-codel] could be used,
but it partitions both latency and bandwdith between every e2e
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flow. So it is rather overkill, which brings disadvantages (see
Section 2.2), not least that thousands of queues are needed when
two are sufficient.
1) Protocol: A host needs to distinguish L4S and Classic packets
with an identifier so that the network can classify them into
their separate treatments. [I-D.briscoe-tsvwg-ecn-l4s-id]
considers various alternative identifiers, and concludes that all
alternatives involve compromises, but the ECT(1) codepoint of the
ECN field is a workable solution.
3) Host: Scalable congestion controls already exist. They solve the
scaling problem with TCP first pointed out in [RFC3649]. The one
used most widely (in controlled environments) is Data Centre TCP
(DCTCP [I-D.ietf-tcpm-dctcp]), which has been implemented and
deployed in Windows Server Editions (since 2012), in Linux and in
FreeBSD. Although DCTCP as-is 'works' well over the public
Internet, most implementations lack certain safety features that
will be necessary once it is used outside controlled environments
like data centres (see later). A similar scalable congestion
control will also need to be transplanted into protocols other
than TCP (SCTP, RTP/RTCP, RMCAT, etc.)
(1) (2)
.-------^------. .--------------^-------------------.
,-(3)-----. ______
; ________ : L4S --------. | |
:|Scalable| : _\ ||___\_| mark |
:| sender | : __________ / / || / |______|\ _________
:|________|\; | |/ --------' ^ \1| |
`---------'\__| IP-ECN | Coupling : \|priority |_\
________ / |Classifier| : /|scheduler| /
|Classic |/ |__________|\ --------. ___:__ / |_________|
| sender | \_\ || | |||___\_| mark/|/
|________| / || | ||| / | drop |
Classic --------' |______|
Figure 1: Components of an L4S Solution: 1) Isolation in separate
network queues; 2) Packet Identification Protocol; and 3) Scalable
Sending Host
1.3. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. In this
document, these words will appear with that interpretation only when
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in ALL CAPS. Lower case uses of these words are not to be
interpreted as carrying RFC-2119 significance.
Classic service: The 'Classic' service is intended for all the
congestion control behaviours that currently co-exist with TCP
Reno (e.g. TCP Cubic, Compound, SCTP, etc).
Low-Latency, Low-Loss and Scalable (L4S) service: The 'L4S' service
is intended for traffic from scalable TCP algorithms such as Data
Centre TCP. But it is also more general--it will allow a set of
congestion controls with similar scaling properties to DCTCP (e.g.
Relentless [Mathis09]) to evolve.
Both Classic and L4S services can cope with a proportion of
unresponsive or less-responsive traffic as well (e.g. DNS, VoIP,
etc).
Scalable Congestion Control: A congestion control where flow rate is
inversely proportional to the level of congestion signals. Then,
as flow rate scales, the number of congestion signals per round
trip remains invariant, maintaining the same degree of control.
For instance, DCTCP averages 2 congestion signals per round-trip
whatever the flow rate.
Classic Congestion Control: A congestion control with a flow rate
compatible with standard TCP Reno [RFC5681]. With Classic
congestion controls, as capacity increases enabling higher flow
rates, the number of round trips between congestion signals
(losses or ECN marks) rises in proportion to the flow rate. So
control of queuing and/or utilization becomes very slack. For
instance, with 1500 B packets and an RTT of 18 ms, as TCP Reno
flow rate increases from 2 to 100 Mb/s the number of round trips
between congestion signals rises proportionately, from 2 to 100.
The default congestion control in Linux (TCP Cubic) is Reno-
compatible for most scenarios expected for some years. For
instance, with a typical domestic round-trip time (RTT) of 18ms,
TCP Cubic only switches out of Reno-compatibility mode once the
flow rate approaches 1 Gb/s. For a typical data centre RTT of 1
ms, the switch-over point is theoretically 1.3 Tb/s. However,
with a less common transcontinental RTT of 100 ms, it only remains
Reno-compatible up to 13 Mb/s. All examples assume 1,500 B
packets.
Classic ECN: The original proposed standard Explicit Congestion
Notification (ECN) protocol [RFC3168], which requires ECN signals
to be treated the same as drops, both when generated in the
network and when responded to by the sender.
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Site: A home, mobile device, small enterprise or campus, where the
network bottleneck is typically the access link to the site. Not
all network arrangements fit this model but it is a useful, widely
applicable generalisation.
1.4. The Standardization Problem
0) Architecture: The first step will be to articulate the structure
and interworking requirements of the set of parts that would
satisfy the overall application performance requirements.
Then specific interworking aspects of the following three components
parts will need to be defined:
1) Protocol:
A. [I-D.briscoe-tsvwg-ecn-l4s-id] recommends ECT(1) is used as
the identifier to classify L4S and Classic packets into their
separate treatments, as required by [RFC4774]. The draft also
points out that the original experimental assignment of this
codepoint as an ECN nonce [RFC3540] needs to be made obsolete
(it was never deployed, and it offers no security benefit now
that deployment is optional).
B. An essential aspect of a scalable congestion control is the
use of explicit congestion signals rather than losses, because
the signals need to be sent immediately and frequently--too
often to use drops. 'Classic' ECN [RFC3168] requires an ECN
signal to be treated the same as a drop, both when it is
generated in the network and when it is responded to by hosts.
L4S allows networks and hosts to support two separate meanings
for ECN. So the standards track [RFC3168] will need to be
updated to allow ECT(1) packets to depart from the 'same as
drop' constraint.
2) Network: The Dual Queue Coupled AQM has been specified as
generically as possible [I-D.briscoe-aqm-dualq-coupled] as a
'semi-permeable' membrane without specifying the particular AQMs
to use in the two queues. An informational appendix of the draft
is provided for pseudocode examples of different possible AQM
approaches. Initially a zero-config variant of RED called Curvy
RED was implemented, tested and documented. A variant of PIE has
been implemented and tested and is about to be documented. The
aim is for designers to be free to implement diverse ideas. So
the brief normative body of the draft only specifies the minimum
constraints an AQM needs to comply with to ensure that the L4S and
Classic services will coexist.
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3) Host:
A. Data Centre TCP is the most widely used example of a scalable
congestion control. It is being documented in the TCPM WG as
an informational record of the protocol currently in use
[I-D.ietf-tcpm-dctcp]. It will be necessary to define a
number of safety features for a variant usable on the public
Internet. A draft list of these, known as the TCP Prague
requirements, has been drawn up (see Appendix A).
B. Transport protocols other than TCP use various congestion
controls designed to be friendly with Classic TCP. It will be
necessary to implement scalable variants of each of these
transport behaviours before they can use the L4S service. The
following standards track RFCs currently define these
protocols, and they will need to be updated to allow a
different congestion response, which they will have to
indicate by using the ECT(1) codepoint: ECN in TCP [RFC3168],
in SCTP [RFC4960], in RTP [RFC6679], and in DCCP [RFC4340].
C. ECN feedback is sufficient for L4S in some transport protocols
(RTCP, DCCP) but not others:
+ For the case of TCP, the feedback protocol for ECN embeds
the assumption from Classic ECN that it is the same as
drop, making it unusable for a scalable TCP. Therefore,
the implementation of TCP receivers will have to be
upgraded [RFC7560]. Work to standardize more accurate ECN
feedback for TCP (AccECN [I-D.ietf-tcpm-accurate-ecn]) is
already in progress.
+ ECN feedback is only roughly sketched in an appendix of the
SCTP specification. A fuller specification has been
proposed [I-D.stewart-tsvwg-sctpecn], which would need to
be implemented and deployed.
Currently, the new specification of the ECN protocol
[I-D.briscoe-tsvwg-ecn-l4s-id] has been written for the experimental
track. Perhaps a better approach would be to make this a standards
track protocol draft that updates the definition of ECT(1) in all the
above standards track RFCs and obsoletes its experimental use for the
ECN nonce. Then experimental specifications of example network (AQM)
and host (congestion control) algorithms can be written.
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2. Rationale
2.1. Why These Primary Components?
Explicit congestion signalling (protocol): Explicit congestion
signalling is a key part of the L4S approach. In contrast, use of
drop as a congestion signal creates a tension because drop is both
a useful signal (more would reduce delay) and an impairment (less
would reduce delay). Explicit congestion signals can be used many
times per round trip, to keep tight control, without any
impairment. Under heavy load, even more explicit signals can be
applied so the queue can be kept short whatever the load. Whereas
state-of-the-art AQMs have to introduce very high packet drop at
high load to keep the queue short. Further, TCP's sawtooth
reduction can be smaller, and therefore return to the operating
point more often, without worrying that this causes more signals
(one at the top of each smaller sawtooth). The consequent smaller
amplitude sawteeth fit between a very shallow marking threshold
and an empty queue, so delay variation can be very low, without
risk of under-utilization.
All the above makes it clear that explicit congestion signalling
is only advantageous for latency if it does not have to be
considered 'the same as' drop (as required with Classic ECN
[RFC3168]). Before Classic ECN was standardized, there were
various proposals to give an ECN mark a different meaning from
drop. However, there was no particular reason to agree on any one
of the alternative meanings, so 'the same as drop' was the only
compromise that could be reached. RFC 3168 contains a statement
that:
"An environment where all end nodes were ECN-Capable could
allow new criteria to be developed for setting the CE
codepoint, and new congestion control mechanisms for end-node
reaction to CE packets. However, this is a research issue, and
as such is not addressed in this document."
Latency isolation with coupled congestion notification (network):
Using just two queues is not essential to L4S (more would be
possible), but it is the simplest way to isolate all the L4S
traffic that keeps latency low from all the legacy Classic traffic
that does not.
Similarly, coupling the congestion notification between the queues
is not necessarily essential, but it is a clever and simple way to
allow senders to determine their rate, packet-by-packet, rather
than be overridden by a network scheduler. Because otherwise a
network scheduler would have to inspect at least transport layer
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headers, and it would have to continually assign a rate to each
flow without any easy way to understand application intent.
L4S packet identifier (protocol): Once there are at least two
separate treatments in the network, hosts need an identifier at
the IP layer to distinguish which treatment they intend to use.
Scalable congestion notification (host): A scalable congestion
control keeps the signalling frequency high so that rate
variations can be small when signalling is stable, and rate can
track variations in available capacity as rapidly as possible
otherwise.
2.2. Why Not Alternative Approaches?
All the following approaches address some part of the same problem
space as L4S. In each case, it is shown that L4S complements them or
improves on them, rather than being a mutually exclusive alternative:
Diffserv: Diffserv addresses the problem of bandwidth apportionment
for important traffic as well as queuing latency for delay-
sensitive traffic. L4S solely addresses the problem of queuing
latency. Diffserv will still be necessary where important traffic
requires priority (e.g. for commercial reasons, or for protection
of critical infrastructure traffic). Nonetheless, if there are
Diffserv classes for important traffic, the L4S approach can
provide low latency for _all_ traffic within each Diffserv class
(including the case where there is only one Diffserv class).
Also, as already explained, Diffserv only works for a small subset
of the traffic on a link. It is not applicable when all the
applications in use at one time at a single site (home, small
business or mobile device) require low latency. Also, because L4S
is for all traffic, it needs none of the management baggage
(traffic policing, traffic contracts) associated with favouring
some packets over others. This baggage has held Diffserv back
from widespread end-to-end deployment.
State-of-the-art AQMs: AQMs such as PIE and fq_CoDel give a
significant reduction in queuing delay relative to no AQM at all.
The L4S work is intended to complement these AQMs, and we
definitely do not want to distract from the need to deploy them as
widely as possible. Nonetheless, without addressing the large
saw-toothing rate variations of Classic congestion controls, AQMs
alone cannot reduce queuing delay too far without significantly
reducing link utilization. The L4S approach resolves this tension
by ensuring hosts can minimize the size of their sawteeth without
appearing so aggressive to legacy flows that they starve.
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Per-flow queuing: Similarly per-flow queuing is not incompatible
with the L4S approach. However, one queue for every flow can be
thought of as overkill compared to the minimum of two queues for
all traffic needed for the L4S approach. The overkill of per-flow
queuing has side-effects:
A. fq makes high performance networking equipment costly
(processing and memory) - in contrast dual queue code can be
very simple;
B. fq requires packet inspection into the end-to-end transport
layer, which doesn't sit well alongside encryption for privacy
- in contrast a dual queue only operates at the IP layer;
C. fq decides packet-by-packet which flow to schedule without
knowing application intent. In contrast, in the L4S approach
the sender still controls the relative rate of each flow
dependent on the needs of each application.
Alternative Back-off ECN (ABE): Yet again, L4S is not an alternative
to ABE but a complement that introduces much lower queuing delay.
ABE [I-D.khademi-tcpm-alternativebackoff-ecn] alters the host
behaviour in response to ECN marking to utilize a link better and
give ECN flows a faster throughput, but it assumes the network
still treats ECN and drop the same. Therefore ABE exploits any
lower queuing delay that AQMs can provide. But as explained
above, AQMs still cannot reduce queuing delay too far without
losing link utilization (for other non-ABE flows).
3. Opportunities
A transport layer that solves the current latency issues will provide
new service, product and application opportunities.
With the L4S approach, the following existing applications will
immediately experience significantly better quality of experience
under load in the best effort class:
o Gaming
o VoIP
o Video conferencing
o Web browsing
o (Adaptive) video streaming
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o Instant messaging
The significantly lower queuing latency also enables some interactive
application functions to be offloaded to the cloud that would hardly
even be usable today:
o Cloud based interactive video
o Cloud based virtual and augmented reality
The above two applications have been successfully demonstrated with
L4S, both running together over a 40 Mb/s broadband access link
loaded up with the numerous other latency sensitive applications in
the previous list as well as numerous downloads. A panoramic video
of a football stadium can be swiped and pinched so that on the fly a
proxy in the cloud generates a sub-window of the match video under
the finger-gesture control of each user. At the same time, a virtual
reality headset fed from a 360 degree camera in a racing car has been
demonstrated, where the user's head movements control the scene
generated in the cloud. In both cases, with 7 ms end-to-end base
delay, the additional queuing delay of roughly 1 ms is so low that it
seems the video is generated locally. See https://riteproject.eu/
dctth/ for videos of these demonstrations.
Using a swiping finger gesture or head movement to pan a video are
extremely demanding applications--far more demanding than VoIP.
Because human vision can detect extremely low delays of the order of
single milliseconds when delay is translated into a visual lag
between a video and a reference point (the finger or the orientation
of the head).
If low network delay is not available, all fine interaction has to be
done locally and therefore much more redundant data has to be
downloaded. When all interactive processing can be done in the
cloud, only the data to be rendered for the end user needs to be
sent. Whereas, once applications can rely on minimal queues in the
network, they can focus on reducing their own latency by only
minimizing the application send queue.
3.1. Use Cases
The following use-cases for L4S are being considered by various
interested parties:
o Where the bottleneck is one of various types of access network:
DSL, cable, mobile, satellite
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* Radio links (cellular, WiFi) that are distant from the source
are particularly challenging. The radio link capacity can vary
rapidly by orders of magnitude, so it is often desirable to
hold a buffer to utilise sudden increases of capacity;
* cellular networks are further complicated by a perceived need
to buffer in order to make hand-overs imperceptible;
* Satellite networks generally have a very large base RTT, so
even with minimal queuing, overall delay can never be extremely
low;
* Nonetheless, it is certainly desirable not to hold a buffer
purely because of the sawteeth of Classic TCP, when it is more
than is needed for all the above reasons.
o Private networks of heterogeneous data centres, where there is no
single administrator that can arrange for all the simultaneous
changes to senders, receivers and network needed to deploy DCTCP:
* a set of private data centres interconnected over a wide area
with separate administrations, but within the same company
* a set of data centres operated by separate companies
interconnected by a community of interest network (e.g. for the
finance sector)
* multi-tenant (cloud) data centres where tenants choose their
operating system stack (Infrastructure as a Service - IaaS)
o Different types of transport (or application) congestion control:
* elastic (TCP/SCTP);
* real-time (RTP, RMCAT);
* query (DNS/LDAP).
o Where low delay quality of service is required, but without
inspecting or intervening above the IP layer
[I-D.you-encrypted-traffic-management]:
* mobile and other networks have tended to inspect higher layers
in order to guess application QoS requirements. However, with
growing demand for support of privacy and encryption, L4S
offers an alternative. There is no need to select which
traffic to favour for queuing, when L4S gives favourable
queuing to all traffic.
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4. IANA Considerations
This specification contains no IANA considerations.
5. Security Considerations
5.1. Traffic (Non-)Policing
Because the L4S service can serve all traffic that is using the
capacity of a link, it should not be necessary to police access to
the L4S service. In contrast, Diffserv only works if some packets
get less favourable treatement than others. So it has to use traffic
policers to limit how much traffic can be favoured, In turn, traffic
policers require traffic contracts between users and networks as well
as pairwise between networks. Because L4S will lack all this
management complexity, it is more likely to work end-to-end.
During early deployment (and perhaps always), some networks will not
offer the L4S service. These networks do not need to police or re-
mark L4S traffic - they just forward it unchanged as best efforts
traffic, as they would already forward traffic with ECT(1) today. At
a bottleneck, such networks will introduce some queuing and dropping.
When a scalable congestion control detects a drop it will have to
respond as if it is a Classic congestion control (see item 3-1 in
Appendix A). This will ensure safe interworking with other traffic
at the 'legacy' bottleneck.
Certain network operators might choose to restict access to the L4S
class, perhaps only to customers who have paid a premium. In the
packet classifer (item 2 in Figure 1), they could identify such
customers using some other field than ECN (e.g. source address
range), and just ignore the L4S identifier for non-paying customers.
This would ensure that the L4S identifier survives end-to-end even
though the service does not have to be supported at every hop. Such
arrangements would only require simple registered/not-registered
packet classification, rather than the managed application-specific
traffic policing against customer-specific traffic contracts that
Diffserv requires.
5.2. 'Latency Friendliness'
The L4S service does rely on self-constraint - not in terms of
limiting capacity usage, but in terms of limiting burstiness. It is
believed that standardisation of dynamic behaviour (cf. TCP slow-
start) and self-interest will be sufficient to prevent transports
from sending excessive bursts of L4S traffic, given the application's
own latency will suffer most from such behaviour.
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Whether burst policing becomes necessary remains to be seen. Without
it, there will be potential for attacks on the low latency of the L4S
service. However it may only be necessary to apply such policing
reactively, e.g. punitively targeted at any deployments of new bursty
malware.
5.3. ECN Integrity
Receiving hosts can fool a sender into downloading faster by
suppressing feedback of ECN marks (or of losses if retransmissions
are not necessary or available otherwise). [RFC3540] proposes that a
TCP sender could pseudorandomly set either of ECT(0) or ECT(1) in
each packet of a flow and remember the sequence it had set, termed
the ECN nonce. If the receiver supports the nonce, it can prove that
it is not suppressing feedback by reflecting its knowledge of the
sequence back to the sender. The nonce was proposed on the
assumption that receivers might be more likely to cheat congestion
control than senders (although senders also have a motive to cheat).
If L4S uses the ECT(1) codepoint of ECN for packet classification, it
will have to obsolete the experimental nonce. As far as is known,
the ECN Nonce has never been deployed, and it was only implemented
for a couple of testbed evaluations. It would be nearly impossible
to deploy now, because any misbehaving receiver can simply opt-out,
which would be unremarkable given all receivers currently opt-out.
Other ways to protect TCP feedback integrity have since been
developed. For instance:
o the sender can test the integrity of the receiver's feedback by
occasionally setting the IP-ECN field to a value normally only set
by the network. Then it can test whether the receiver's feedback
faithfully reports what it expects [I-D.moncaster-tcpm-rcv-cheat].
This method consumes no extra codepoints. It works for loss and
it will work for ECN feedback in any transport protocol suitable
for L4S. However, it shares the same assumption as the nonce;
that the sender is not cheating and it is motivated to prevent the
receiver cheating;
o A network can enforce a congestion response to its ECN markings
(or packet losses) by auditing congestion exposure (ConEx)
[RFC7713]. Whether the receiver or a downstream network is
suppressing congestion feedback or the sender is unresponsive to
the feedback, or both, ConEx audit can neutralise any advantage
that any of these three parties would otherwise gain. ConEx is
only currently defined for IPv6 and consumes a destination option
header. It has been implemented, but not deployed as far as is
known.
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6. Acknowledgements
7. References
7.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
7.2. Informative References
[Alizadeh-stability]
Alizadeh, M., Javanmard, A., and B. Prabhakar, "Analysis
of DCTCP: Stability, Convergence, and Fairness", ACM
SIGMETRICS 2011 , June 2011.
[DCttH15] De Schepper, K., Bondarenko, O., Briscoe, B., and I.
Tsang, "'Data Centre to the Home': Ultra-Low Latency for
All", 2015, <http://www.bobbriscoe.net/projects/latency/
dctth_preprint.pdf>.
(Under submission)
[Hohlfeld14]
Hohlfeld , O., Pujol, E., Ciucu, F., Feldmann, A., and P.
Barford, "A QoE Perspective on Sizing Network Buffers",
Proc. ACM Internet Measurement Conf (IMC'14) hmm, November
2014.
[I-D.briscoe-aqm-dualq-coupled]
Schepper, K., Briscoe, B., Bondarenko, O., and I. Tsang,
"DualQ Coupled AQM for Low Latency, Low Loss and Scalable
Throughput", draft-briscoe-aqm-dualq-coupled-01 (work in
progress), March 2016.
[I-D.briscoe-tsvwg-ecn-l4s-id]
Schepper, K., Briscoe, B., and I. Tsang, "Identifying
Modified Explicit Congestion Notification (ECN) Semantics
for Ultra-Low Queuing Delay", draft-briscoe-tsvwg-ecn-l4s-
id-01 (work in progress), March 2016.
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[I-D.ietf-aqm-fq-codel]
Hoeiland-Joergensen, T., McKenney, P.,
dave.taht@gmail.com, d., Gettys, J., and E. Dumazet, "The
FlowQueue-CoDel Packet Scheduler and Active Queue
Management Algorithm", draft-ietf-aqm-fq-codel-06 (work in
progress), March 2016.
[I-D.ietf-aqm-pie]
Pan, R., Natarajan, P., Baker, F., and G. White, "PIE: A
Lightweight Control Scheme To Address the Bufferbloat
Problem", draft-ietf-aqm-pie-08 (work in progress), June
2016.
[I-D.ietf-tcpm-accurate-ecn]
Briscoe, B., Kuehlewind, M., and R. Scheffenegger, "More
Accurate ECN Feedback in TCP", draft-ietf-tcpm-accurate-
ecn-01 (work in progress), June 2016.
[I-D.ietf-tcpm-cubic]
Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
draft-ietf-tcpm-cubic-01 (work in progress), January 2016.
[I-D.ietf-tcpm-dctcp]
Bensley, S., Eggert, L., Thaler, D., Balasubramanian, P.,
and G. Judd, "Datacenter TCP (DCTCP): TCP Congestion
Control for Datacenters", draft-ietf-tcpm-dctcp-01 (work
in progress), November 2015.
[I-D.khademi-tcpm-alternativebackoff-ecn]
Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
"TCP Alternative Backoff with ECN (ABE)", draft-khademi-
tcpm-alternativebackoff-ecn-00 (work in progress), May
2016.
[I-D.moncaster-tcpm-rcv-cheat]
Moncaster, T., Briscoe, B., and A. Jacquet, "A TCP Test to
Allow Senders to Identify Receiver Non-Compliance", draft-
moncaster-tcpm-rcv-cheat-03 (work in progress), July 2014.
[I-D.stewart-tsvwg-sctpecn]
Stewart, R., Tuexen, M., and X. Dong, "ECN for Stream
Control Transmission Protocol (SCTP)", draft-stewart-
tsvwg-sctpecn-05 (work in progress), January 2014.
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[I-D.you-encrypted-traffic-management]
You, J. and C. Xiong, "The Effect of Encrypted Traffic on
the QoS Mechanisms in Cellular Networks", draft-you-
encrypted-traffic-management-00 (work in progress),
October 2015.
[Mathis09]
Mathis, M., "Relentless Congestion Control", PFLDNeT'09 ,
May 2009, <http://www.hpcc.jp/pfldnet2009/
Program_files/1569198525.pdf>.
[NewCC_Proc]
Eggert, L., "Experimental Specification of New Congestion
Control Algorithms", IETF Operational Note ion-tsv-alt-cc,
July 2007.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<http://www.rfc-editor.org/info/rfc3168>.
[RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec,
J., Courtney, W., Davari, S., Firoiu, V., and D.
Stiliadis, "An Expedited Forwarding PHB (Per-Hop
Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002,
<http://www.rfc-editor.org/info/rfc3246>.
[RFC3540] Spring, N., Wetherall, D., and D. Ely, "Robust Explicit
Congestion Notification (ECN) Signaling with Nonces",
RFC 3540, DOI 10.17487/RFC3540, June 2003,
<http://www.rfc-editor.org/info/rfc3540>.
[RFC3649] Floyd, S., "HighSpeed TCP for Large Congestion Windows",
RFC 3649, DOI 10.17487/RFC3649, December 2003,
<http://www.rfc-editor.org/info/rfc3649>.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340,
DOI 10.17487/RFC4340, March 2006,
<http://www.rfc-editor.org/info/rfc4340>.
[RFC4774] Floyd, S., "Specifying Alternate Semantics for the
Explicit Congestion Notification (ECN) Field", BCP 124,
RFC 4774, DOI 10.17487/RFC4774, November 2006,
<http://www.rfc-editor.org/info/rfc4774>.
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[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007,
<http://www.rfc-editor.org/info/rfc4960>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<http://www.rfc-editor.org/info/rfc5681>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <http://www.rfc-editor.org/info/rfc6679>.
[RFC7560] Kuehlewind, M., Ed., Scheffenegger, R., and B. Briscoe,
"Problem Statement and Requirements for Increased Accuracy
in Explicit Congestion Notification (ECN) Feedback",
RFC 7560, DOI 10.17487/RFC7560, August 2015,
<http://www.rfc-editor.org/info/rfc7560>.
[RFC7713] Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx)
Concepts, Abstract Mechanism, and Requirements", RFC 7713,
DOI 10.17487/RFC7713, December 2015,
<http://www.rfc-editor.org/info/rfc7713>.
[TCP-sub-mss-w]
Briscoe, B. and K. De Schepper, "Scaling TCP's Congestion
Window for Small Round Trip Times", BT Technical Report
TR-TUB8-2015-002, May 2015,
<http://www.bobbriscoe.net/projects/latency/
sub-mss-w.pdf>.
[TCPPrague]
Briscoe, B., "Notes: DCTCP evolution 'bar BoF': Tue 21 Jul
2015, 17:40, Prague", tcpprague mailing list archive ,
July 2015.
Appendix A. Required features for scalable transport protocols to be
safely deployable in the Internet (a.k.a. TCP Prague
requirements)
This list contains a list of features, mechanisms and modifications
from currently defined behaviour for scalable Transport protocols so
that they can be safely deployed over the public Internet. This list
of requirements was produced at an ad hoc meeting during IETF-94 in
Prague [TCPPrague].
One of such scalable transport protocols is DCTCP, currently
specified in [I-D.ietf-tcpm-dctcp]. In its current form, DCTCP is
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specified to be deployable in controlled environments and deploying
it in the public Internet would lead to a number of issues, both from
the safety and the performance perspective. In this section, we
describe the modifications and additional mechanisms that are
required for its deployment over the global Internet. We use DCTCP
as a base, but it is likely that most of these requirements equally
apply to other scalable transport protocols.
We next provide a brief description of each required feature.
Requirement #4.1: Fall back to Reno/Cubic congestion control on
packet loss.
Description: In case of packet loss, the scalable transport MUST
react as classic TCP (whatever the classic version of TCP is running
in the host, e.g. Reno, Cubic).
Motivation: As part of the safety conditions for deploying a scalable
transport over the public Internet is to make sure that it behaves
properly when some or all the network devices connecting the two
endpoints that implement the scalable transport have not been
upgraded. In particular, it may be the case that some of the
switches along the path between the two endpoints may only react to
congestion by dropping packets (i.e. no ECN marking). It is
important that in these cases, the scalable transport react to the
congestion signal in the form of a packet drop similarly to classic
TCP.
In the particular case of DCTCP, the current DCTCP specification
states that "It is RECOMMENDED that an implementation deal with loss
episodes in the same way as conventional TCP." For safe deployment
in the public Internet of a scalable transport, the above requirement
needs to be defined as a MUST.
Packet loss, while rare, may also occur in the case that the
bottleneck is L4S capable. In this case, the sender may receive a
high number of packets marked with the CE bit set and also experience
a loss. Current DCTCP implementations react differently to this
situation. At least one implementation reacts only to the drop
signal (e.g. by halving the CWND) and at least another DCTCP
implementation reacts to both signals (e.g. by halving the CWND due
to the drop and also further reducing the CWND based on the
proportion of marked packet). We believe that further
experimentation is needed to understand what is the best behaviour
for the public Internet, which may or not be one of the existent
implementations.
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Requirement #4.2: Fall back to Reno/Cubic congestion control on
classic ECN bottlenecks.
Description: The scalable transport protocol SHOULD/MAY? behave as
classic TCP with classic ECN if the path contains a legacy bottleneck
which marks both ect(0) and ect(1) in the same way as drop (non L4S,
but ECN capable bottleneck).
Motivation: Similarly to Requirement #3.1, this requirement is a
safety condition in case L4S-capable endpoints are communicating over
a path that contains one or more non-L4S but ECN capable switches and
one of them happens to be the bottleneck. In this case, the scalable
transport will attempt to fill in the buffer of the bottleneck switch
up to the marking threshold and produce a small sawtooth around that
operation point. The result is that the switch will set its
operation point with the buffer full and all other non-scalable
transports will be starved (as they will react reducing their CWND
more aggressively than the scalable transport).
Scalable transports then MUST be able to detect the presence of a
classic ECN bottleneck and fall back to classic TCP/classic ECN
behaviour in this case.
Discussion: It is not clear at this point if it is possible to design
a mechanism that always detect the aforementioned cases. One
possibility is to base the detection on an increase on top of a
minimum RTT, but it is not yet clear which value should trigger this.
Having a delay based fall back response on L4S may as well be
beneficial for preserving low latency without legacy network nodes.
Even if it possible to design such a mechanism, it may well be that
it would encompass additional complexity that implementers may
consider unnecessary. The need for this mechanism depends on the
extent of classic ECN deployment.
Requirement #4.3: Reduce RTT dependence
Description: Scalable transport congestion control algorithms MUST
reduce or eliminate the RTT bias within the range of RTTs available.
Motivation: Classic TCP's throughput is known to be inversely
proportional to RTT. One would expect flows over very low RTT paths
to nearly starve flows over larger RTTs. However, because Classic
TCP induces a large queue, it has never allowed a very low RTT path
to exist, so far. For instance, consider two paths with base RTT 1ms
and 100ms. If Classic TCP induces a 20ms queue, it turns these RTTs
into 21ms and 120ms leading to a throughput ratio of about 1:6.
Whereas if a Scalable TCP induces only a 1ms queue, the ratio is
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2:101. Therefore, with small queues, long RTT flows will essentially
starve.
Scalable transport protocol MUST then accommodate flows across the
range of RTTs enabled by the deployment of L4S service over the
public Internet.
Requirement #4.4: Scaling down the congestion window.
Description: Scalable transports MUST be responsive to congestion
when RTTs are significantly smaller than in the current public
Internet.
Motivation: As currently specified, the minimum CWND of TCP (and the
scalable extensions such as DCTCP), is set to 2 MSS. Once this
minimum CWND is reached, the transport protocol ceases to react to
congestion signals (the CWND is not further reduced beyond this
minimum size).
L4S mechanisms reduce significantly the queueing delay, achieving
smaller RTTs over the Internet. For the same CWND, smaller RTTs
imply higher transmission rates. The result is that when scalable
transport are used and small RTTs are achieved, the minimum value of
the CWND currently defined in 2 MSS may still result in a high
transmission rate for a large number of common scenarios. For
example, as described in [TCP-sub-mss-w], consider a residential
setting with an broadband Internet access of 40Mbps. Suppose now a
number of equal TCP flows running in parallel with the Internet
access link being the bottleneck. Suppose that for these flows, the
RTT is 6ms and the MSS is 1500B. The minimum transmission rate
supported by TCP in this scenario is when CWND is set to 2 MSS, which
results in 4Mbps for each flow. This means that in this scenario, if
the number of flows is higher than 10, the congestion control ceases
to be responsive and starts to build up a queue in the network.
In order to address this issue, the congestion control mechanism for
scalable transports MUST be responsive for the new range of RTT
resulting from the decrease of the queueing delay.
There are several ways how this can be achieved. One possible sub-
MSS window mechanism is described in [TCP-sub-mss-w].
In addition to the safety requirements described before, there are
some optimizations that while not required for the safe deployment of
scalable transports over the public Internet, would results in an
optimized performance. We describe them next.
Optimization #5.1: Setting ECT in SYN, SYN/ACK and pure ACK packets.
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Description: Scalable transport SHOULD set the ECT bit in SYN, SYN/
ACK and pure ACK packets.
Motivation: Failing to set the ECT bit in SYN, SYN/ACK or ACK packets
results in these packets being more likely dropped during congestion
events. Dropping SYN and SYN/ACK packets is particularly bad for
performance as the retransmission timers for these packets are large.
[RFC3168] prevents from marking these packets due to security
reasons. The arguments provided should be revisited in the the
context of L4S and evaluate if avoiding marking these packets is
still the best approach.
Optimization #5.2: Faster than additive increase.
Description: Scalable transport MAY support faster than additive
increase in the congestion avoidance phase.
Motivation: As currently defined, DCTCP supports additive increase in
congestion avoidance phase. It would be beneficial for performance
to update the congestion control algorithm to increase the CWND more
than 1 MSS per RTT during the congestion avoidance phase. In the
context of L4S such mechanism, must also provide fairness with other
classes of traffic, including classic TCP and possibly scalable TCP
that uses additive increase.
Optimization #5.3: Faster convergence to fairness.
Description: Scalable transport SHOULD converge to a fair share
allocation of the available capacity as fast as classic TCP or
faster.
Motivation: The time required for a new flow to obtain its fair share
of the capacity of the bottleneck when the there are already ongoing
flows using up all the bottleneck capacity is higher in the case of
DCTCP than in the case of classic TCP (about a factor of 1,5 and 2
larger according to [Alizadeh-stability]). This is detrimental in
general, but it is very harmful for short flows, which performance
can be worse than the one obtained with classic TCP. for this reason
it is desirable that scalable transport provide convergence times no
larger than classic TCP.
Appendix B. Standardization items
The following table includes all the itmes that should be
standardized to provide a full L4S architecture.
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The table is too wide for the ASCII draft format, so it has been
split into two, with a common column of row index numbers on the
left.
The columns in the second part of the table have the following
meanings:
WG: The IETF WG most relevant to this requirement. The "tcpm/iccrg"
combination refers to the procedure typically used for congestion
control changes, where tcpm owns the approval decision, but uses
the iccrg for expert review [NewCC_Proc];
TCP: Applicable to all forms of TCP congestion control;
DCTCP: Applicable to Data Centre TCP as currently used (in
controlled environments);
DCTCP bis: Applicable to an future Data Centre TCP congestion
control intended for controlled environments;
XXX Prague: Applicable to a Scalable variant of XXX (TCP/SCTP/RMCAT)
congestion control.
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+-----+-----------------------+-------------------------------------+
| Req | Requirement | Reference |
| # | | |
+-----+-----------------------+-------------------------------------+
| 0 | ARCHITECTURE | |
| 1 | L4S IDENTIFIER | [I-D.briscoe-tsvwg-ecn-l4s-id] |
| 2 | DUAL QUEUE AQM | [I-D.briscoe-aqm-dualq-coupled] |
| 3 | Suitable ECN Feedback | [I-D.ietf-tcpm-accurate-ecn], |
| | | [I-D.stewart-tsvwg-sctpecn]. |
| | | |
| | SCALABLE TRANSPORT - | |
| | SAFETY ADDITIONS | |
| 4-1 | Fall back to | [I-D.ietf-tcpm-dctcp] |
| | Reno/Cubic on loss | |
| 4-2 | Fall back to | |
| | Reno/Cubic if classic | |
| | ECN bottleneck | |
| | detected | |
| | | |
| 4-3 | Reduce RTT-dependence | |
| | | |
| 4-4 | Scaling TCP's | [TCP-sub-mss-w] |
| | Congestion Window for | |
| | Small Round Trip | |
| | Times | |
| | SCALABLE TRANSPORT - | |
| | PERFORMANCE | |
| | ENHANCEMENTS | |
| 5-1 | Setting ECT in SYN, | draft-bagnulo-tsvwg-generalized-ECN |
| | SYN/ACK and pure ACK | |
| | packets | |
| 5-2 | Faster-than-additive | |
| | increase | |
| 5-3 | Less drastic exit | |
| | from slow-start | |
+-----+-----------------------+-------------------------------------+
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+-----+--------+-----+-------+-----------+--------+--------+--------+
| # | WG | TCP | DCTCP | DCTCP-bis | TCP | SCTP | RMCAT |
| | | | | | Prague | Prague | Prague |
+-----+--------+-----+-------+-----------+--------+--------+--------+
| 0 | tsvwg? | Y | Y | Y | Y | Y | Y |
| 1 | tsvwg? | | | Y | Y | Y | Y |
| 2 | aqm? | n/a | n/a | n/a | n/a | n/a | n/a |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| 3 | tcpm | Y | Y | Y | Y | n/a | n/a |
| | | | | | | | |
| 4-1 | tcpm | | Y | Y | Y | Y | Y |
| | | | | | | | |
| 4-2 | tcpm/ | | | | Y | Y | ? |
| | iccrg? | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| | | | | | | | |
| 4-3 | tcpm/ | | | Y | Y | Y | ? |
| | iccrg? | | | | | | |
| 4-4 | tcpm | Y | Y | Y | Y | Y | ? |
| | | | | | | | |
| | | | | | | | |
| 5-1 | tsvwg | Y | Y | Y | Y | n/a | n/a |
| | | | | | | | |
| 5-2 | tcpm/ | | | Y | Y | Y | ? |
| | iccrg? | | | | | | |
| 5-3 | tcpm/ | | | Y | Y | Y | ? |
| | iccrg? | | | | | | |
+-----+--------+-----+-------+-----------+--------+--------+--------+
Authors' Addresses
Bob Briscoe (editor)
Simula Research Lab
Email: ietf@bobbriscoe.net
URI: http://bobbriscoe.net/
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Koen De Schepper
Nokia Bell Labs
Antwerp
Belgium
Email: koen.de_schepper@nokia.com
URI: https://www.bell-labs.com/usr/koen.de_schepper
Marcelo Bagnulo
Universidad Carlos III de Madrid
Av. Universidad 30
Leganes, Madrid 28911
Spain
Phone: 34 91 6249500
Email: marcelo@it.uc3m.es
URI: http://www.it.uc3m.es
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