Internet Engineering Task Force G. Montenegro
INTERNET DRAFT Sun Microsystems, Inc.
S. Dawkins
Nortel Networks
M. Kojo
University of Helsinki
V. Magret
Alcatel
November 18, 1998
Long Thin Networks
draft-montenegro-pilc-ltn-00.txt
Status of This Memo
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Abstract
In view of the unpredictable and problematic nature of long thin
networks (for example, wireless WANs), arriving at an optimized
transport is a daunting task. We have reviewed the existing
proposals along with future research items. Based on this
overview, we also recommend mechanisms for implementation in long
thin networks.
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Our goal is to identify a TCP that works for all users, including
users of long thin networks. We started from the working
recommendations of the IETF TCP Over Satellite Links (tcpsat)
working group with this end in mind.
We recognize that not every tcpsat recommendation will be required
for long thin networks as well, and are working toward a set of
TCP recommendations that are 'benign' in environments that do not
require them.
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Table of Contents
1 Introduction .................................................... 4
1.1 Architecture ............................................... 6
1.2 Assumptions about the Radio Link ........................... 7
2 Should it be IP or Not? ........................................ 8
2.1 Underlying Network Error Characteristics ................... 8
2.2 Non-IP Alternatives ........................................ 9
2.2.1 WAP ................................................... 9
2.2.2 Deploying Non-IP Alternatives ......................... 10
2.3 IP-based Alternatives ...................................... 10
2.3.1 Path MTU Discovery .................................... 10
2.3.2 Non-TCP Proposals ..................................... 11
3 The Case for TCP ................................................ 11
4 Candidate Optimizations ......................................... 12
4.1 TCP: Current Mechanisms .................................... 13
4.1.1 Slow Start and Congestion Avoidance ................... 13
4.1.2 Fast Retransmit and Fast Recovery ..................... 13
4.2 Connection Setup with T/TCP [RFC1397, RFC1644] ............. 14
4.3 Slow Start Proposals ....................................... 14
4.3.1 Larger Initial Window ................................. 14
4.3.2 Handling Acknowledgments During Slow Start ............ 15
4.3.2.1 ACK Counting ..................................... 15
4.3.2.2 ACK-every-segment ................................ 16
4.3.3 Terminating Slow Start ................................ 17
4.4 ACK Spacing ................................................ 17
4.5 Delayed Duplicate Acknowlegements .......................... 18
4.6 Selective Acknowledgements [RFC2018] ....................... 18
4.7 Detecting Corruption Loss .................................. 19
4.7.1 Without Explicit Notification ......................... 19
4.7.2 With Explicit Notifications ........................... 19
4.8 Active Queue Management .................................... 20
4.9 Scheduling Algorithms ...................................... 21
4.10 Split TCP and Performance-Enhancing Proxies (PEPs) ........ 22
4.11 Header Compression Alternatives ........................... 24
4.12 IP Payload Compression .................................... 25
4.13 TCP Control Block Interdependence [Touch97] ............... 26
5 Summary of Recommended Optimizations ............................ 26
6 Conclusion ...................................................... 29
7 Acknowledgements ................................................ 29
8 References ...................................................... 29
Authors' addresses ................................................ 35
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1 Introduction
Optimized wireless networking is one of the major hurdles that
Mobile Computing must solve if it is to enable ubiquitous access
to networking resources. However, current data networking
protocols have been optimized primarily for wired networks.
Wireless environments have very different characteristics in terms
of latency, jitter, and error rate as compared to wired networks.
Accordingly, traditional protocols are ill-suited to this medium.
Mobile Wireless networks can be grouped in W-LANs (for example,
802.11 compliant networks) and W-WANs (for example, CDPD [CDPD],
Ricochet, CDMA [CDMA], PHS, DoCoMo, GSM [GSM] to name a few).
W-WANs present the most serious challenge, given that the length
of the wireless link (expressed as the delay*bandwidth product) is
typically 4 to 5 times as long as that of its W-LAN counterparts.
For example, for an 802.11 network, assuming the delay (round-trip
time) is about 3 ms. and the bandwidth is 1.5 Mbps, the
delay*bandwidth product is 4500 bits. For a W-WAN such as
Ricochet, a typical round-trip time may be around 500 ms. (the
best is about 230 ms.), and the sustained bandwidth is about 24
Kbps. This yields a delay*bandwidth product roughly equal to 1.5
KB. In the near future, 3rd Generation wireless services will
offer 384Kbps and more. Assuming a 200 ms round-trip, the
delay*bandwidth product in this case is 76.8 Kbits (9.6 KB). This
value is larger than the default 8KB buffer space used by many TCP
implementations. This means that, whereas for W-LANs the default
buffer space is enough, future W-WANs will operate inefficiently
(that is, they will not be able to fill the pipe) unless they
override the default value. A 3rd Generation wireless service
offering 2 Mbps with 200-millisecond latency requires a 50 KB
buffer.
Most importantly, latency across a link adversely affects
throughput. For example, [MSMO97] derives an upper bound on TCP
throughput. Indeed, the resultant expression is inversely related
to the round-trip time.
The long latencies also push the limits (and commonly transgress
them) for what is acceptable to users of interactive
applications.
As a quick glance to our list of references will reveal, there is
a wealth of proposals that attempt to solve the wireless
networking problem. In this document, we survey the different
solutions available or under investigation, and issue the
corresponding recommendations.
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There is a large body of work on the subject of improving TCP
performance over satellite links. The documents under development
by the tcpsat working group of the IETF [AGS98, ADGGHOSSTT98] are
very relevant. In both cases, it is essential to start by
improving the characteristics of the medium by using forward error
correction (FEC) at the link layer to reduce the BER (bit error
rate) from values as high as 10-3 to 10-6 or better. This makes
the BER manageable. Once in this realm, retransmission schemes
like ARQ (automatic repeat request) may be used to bring it down
to zero. Notice that sometimes it may be desireable to forgo ARQ
because of the additional delay it implies. In particular, time
sensitive traffic (video, audio) must be delivered within a
certain time limit beyond which the data is obsolete. Exhaustive
retransmissions in this case merely succeed in wasting time in
order to deliver data that will be discarded once it arrives at
its destination. This indicates the desireability of augmenting
the protocol stack implementation on devices such that the upper
protocol layers can inform the link and MAC layer when to avoid
such costly retransmission schemes.
Networks that include satellite links are examples of "long fat
networks" (LFNs or "elephants"). They are "long" networks because
their round-trip time is quite high (for example, 0.5 sec and
higher for geosynchronous satellites). Not all satellite links
fall within the LFN regime. In particular, round-trip times in a
low-earth orbiting (LEO) satellite network may be as little as a
few milliseconds (and never extend beyond 160 to 200 ms). W-WANs
share the "L" with LFNs. However, satellite networks are also
"fat" in the sense that they may have high bandwidth. Satellite
networks may often have a delay*bandwidth product above 64 KBytes,
in which case they pose additional problems to TCP [TCPHP]. W-WANs
do not generally exhibit this behavior. Accordingly, this document
only deals with links that are "long thin pipes", and the networks
that contain them: "long thin networks". We call these "LTNs".
This document does not give an overview of the API used to access
the underlying transport. We believe this is an orthogonal issue,
even though some of the proposals below have been put forth
assuming a given interface. It is possible, for example, to
support the traditional socket semantics without fully relying on
TCP/IP transport [MOWGLI].
Our focus is on the on-the-wire protocols. We try to include the
most relevant ones and briefly (given that we provide the
references needed for further study) mention their most salient
points.
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1.1 Architecture
One significant difference between LFNs and LTNs is that we assume
the W-WAN link is the last hop to the end user. This allows us to
assume that a single base station sees all packets transferred
between the wireless mobile device and the rest of the Internet.
This is only one of the topologies considered by the TCP Satellite
community.
Given our focus on mobile wireless applications, we only consider
a very specific architecture that includes:
- a wireless mobile device, connected via
- a wireless link (which may, in fact comprise several hops at
the link layer), to
- a base station (sometimes referred to as an intermediate
agent) connected via
- a wireline link, which in turn interfaces with
- the landline Internet and millions of legacy servers and web
sites.
Specifically, we are not as concerned with paths that include two
wireless segments separated by a wired one. This may occur, for
example, if one mobile device connects across its immediate
wireless segment via a base station to the internet, and then via
a second wireless segment to another mobile device. Quite often,
mobile devices connect to a legacy server on the wired internet.
Typically, the endpoints of the wireless segment are the base
station and the mobile device. However, the latter may be a
wireless router to a mobile network. This is also important and
has applications in, for example, disaster recovery.
Our target architecture has implications which concern the
deployability of candidate solutions. In particular, an important
requirement is that we cannot alter the networking stack on the
legacy servers. It would be preferable to only change the
networking stack at the base station, although changing it at the
mobile devices is certainly an option and perhaps a necessity.
We envision mobile devices that can use the wireless medium very
efficiently, but overcome some of its traditional constraints.
That is, full mobility implies that the devices have the
flexibility and agility to use whichever happens to be the best
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network connection available at any given point in time or space.
Accordingly, devices could switch from a wired office LAN and hand
over their ongoing connections to continue on, say, a wireless
WAN. This type of agility also requires Mobile IP [RFC2002].
NOTE: Must we replace "base station" with some other term (e.g.,
LTN-edge device)? "Base station" is a good term when discussing
W-LANs but a misleading one in almost all W-WAN environments.
W-LANs, the wireless link is between mobile device and base
station, but within a typical W-WAN infrastructure there are
several wireline hops between the actual W-WAN base station and
the W-WAN edge device that provides the connection point to the
landline Internet. These "base stations" do not have an IP stack,
so, for example, they cannot have a SNOOP module.
1.2 Assumptions about the Radio Link
The system architecture described above assumes at most one
wireless link (perhaps comprising more than one wireless hop).
However, this is not enough to characterize a wireless link.
Additional considerations are:
- What are the error characteristics of the wireless medium?
The link may present a higher BER than a wireline network due
to burst errors and disconnections. The techniques below
usually do not address all the types of errors. Accordingly,
a complete solution should combine the best of all the
proposals. Nevertheless, in this document we are more
concerned with (and give preference to solving) the most
typical case: (1) higher BER due to random errors (which
implies longer and more variable delays due to link-layer
error corrections and retransmissions) rather than (2) an
interruption in service due to a handoff or a disconnection.
The latter are also important and we do include relevant
proposals in this survey.
- Is the wireless service datagram oriented, or is it a virtual
circuit? Currently, switched virtual circuits are more
common, but packet networks are starting to appear, for
example, Metricom's Starmode [CB96], CDPD [CDPD] and General
Packet Radio Service (GPRS) [GPRS],[BW97] in GSM.
- What kind of reliability does the link provide? Wireless
services typically retransmit a packet until it has been
acknowledged by the target. They may allow the user to turn
off this behavior. For example, GSM allows RLP (Reliable Link
Protocol) to be turned off. Metricom has a similar
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"lightweight" mode.
- Does the mobile device transmit and receive at the same
time? Doing so increases the cost of the electronics on the
mobile device. Typically, this is not the case. We assume the
typical case in this document.
- Does the mobile device directly address more than one peer on
the wireless link? Packets to each different peer may
traverse spatially distinct wireless paths. Accordingly, the
path to each peer may exhibit very different
characteristics. Quite commonly, the mobile device addresses
only one peer (the base station) at any given point in time.
When this is not the case, techniques such as Channel-State
Dependent Packet Scheduling come into play (see the section
"Packet Scheduling" below).
2 Should it be IP or Not?
The first decision is whether to use IP as the underlying network
protocol or not. In particular, some data protocols evolved from
wireless telephony are not always -- though at times they may be
-- layered on top of IP [MOWGLI, WAP]. These proposals are based
on the concept of proxies that provide adaptation services between
the wireless and wireline segments.
This is a reasonable model for mobile devices that always
communicate through the proxy. However, we expect many wireless
mobile devices to utilize wireline networks whenever they are
available. This model closely follows current laptop usage
patterns: devices typically utilize LANs, and only resort to
dial-up access when "out of the office."
For these devices, an architecture that assumes IP is the best
approach, because it will be required for communications that do
not traverse the base station (for example, upon reconnection to a
W-LAN or a 10BaseT network at the office).
2.1 Underlying Network Error Characteristics
Using IP as the underlying network protocol requires a certain
(low) level of link robustness that is expected of wireless
links.
IP, and the protocols that are carried in IP packets, are
protected end-to-end by checksums that are relatively weak (and,
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in some cases, optional). For much of the internet, these
checksums are sufficient; in wireless environments, the error
characteristics of the raw wireless link are much less robust than
the rest of the end-to-end path. This makes end-to-end detection
of network errors undesirable, because damaged IP packets are
propagated through the network only to be discarded at the
destination host, and suggests that link-level mechanisms should
be used to detect and correct transmission errors over these
links.
A better approach is to use link-layer mechanisms such as FEC,
retransmissions, and so on in order to improve the characteristics
of the wireless link and present a much more reliable service to
IP. This approach has been taken by CDPD, Ricochet and CDMA.
This approach is roughly analogous to the successful deployment of
Point-to-Point Protocol (PPP), with robust framing and 16-bit
checksumming, on wireline networks as a replacement for the Serial
Line Interface Protocol (SLIP), with only a single framing byte
and no checksumming.
[AGS98] recommends the use of FEC in satellite environments.
Notice that the link-layer could adapt its frame size to the
prevalent BER. It would perform its own fragmentation and
reassembly so that IP could still enjoy a large enough MTU size
[LS98].
A common concern for using IP as a transport is the header
overhead it implies. Typically, the underlying link-layer appears
as PPP [RFC1661] to the IP layer above. This allows for header
compression schemes [IPHC, IPHC-PPP] which greatly alleviate the
problem.
2.2 Non-IP Alternatives
A number of non-IP alternatives aimed at wireless environments
have been proposed. One representative proposal is discussed
here.
2.2.1 WAP
The Wireless Application Protocol (WAP) specifies an application
framework and network protocols for wireless devices such as
mobile telephones, pagers, and PDAs [WAP]. The architecture
requires a proxy between the mobile device and the server. The WAP
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protocol stack is layered over a datagram transport service. Such
a service is provided by most wireless networks; for example,
IS-136, GSM SMS/USSD, and UDP in IP networks like CDPD and GSM
GPRS. The core of the WAP protocols is a binary HTTP/1.1 protocol
with additional features such as header caching between requests
and a shared state between client and server.
2.2.2 Deploying Non-IP Alternatives
IP is such a fundamental element of the internet that non-IP
alternatives face substantial obstacles to deployment, because
they do not exploit the IP infrastructure. Any non-IP alternative
that is used to provide gatewayed access to the internet must map
between IP addresses and non-IP addresses, must terminate IP-level
security at a gateway, and cannot use IP-oriented discovery
protocols (Dynamic Host Configuration Protocol, Domain Name
Services, Lightweight Directory Access Protocol, Service Location
Protocol, etc.) without translation at a gateway.
A further complexity occurs when a device supports both wireless
and wireline operation. If the device uses IP for wireless
operation, uninterrupted operation when the device is connected to
a wireline network is possible (using Mobile IP). If a non-IP
alternative is used, this switchover is more difficult to
accomplish.
Non-IP alternatives face the burden of proof that IP is so
ill-suited to a wireless environment that it is not a viable
technology.
2.3 IP-based Alternatives
Given its worldwide deployment, IP is an obvious choice for the
underlying network technology. Optimizations implemented at this
level benefit traditional internet application protocols as well
as new ones layered on top of IP or UDP.
2.3.1 Path MTU Discovery
Path MTU discovery benefits any protocol built on top of IP. It
allows a sender to determine what the maximum end-to-end
transmission unit is to a given destination. Withouth Path MTU
discovery, the default MTU size is 512. The benefits of using a
larger MTU are:
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- Smaller ratio of header overhead to data
- Allows TCP to grow its congestion window faster, since it
increases in units of segments.
Of course, for a given BER, a larger MTU has a correspondingly
larger probability of error within any given segment. The BER may
be reduced using lower level techniques like FEC and link-layer
retransmissions. The issue is that now delays may become a problem
due to the additional retransmissions, and the fact that packet
transmission time increases with a larger MTU.
[AGS98] recommends use of Path MTU Discovery in satellite
environments.
2.3.2 Non-TCP Proposals
Other proposals assume an underlying IP datagram service, and
implement an optimized transport either directly on top of IP
[NETBLT] or on top of UDP [MNCP]. Not relying on TCP is a bold
move, given the wealth of experience and research related to it.
It could be argued that the internet has not collapsed because its
main protocol, TCP, is very careful in how it uses the network,
and generally treats it as a black box assuming all packet losses
are due to congestion and prudently backing off. This avoids
further congestion.
However, in the wireless medium, packet losses may also be due to
corruption due to high BER, fading, and so on. Here, the right
approach is to try harder, instead of backing off. Alternative
transport protocols are:
- NETBLT [NETBLT, RFC1986, RFC1030]
- MNCP [MNCP]
- ESRO [RFC2188]
- RDP [RFC908, RFC1151]
- VMTP [VMTP]
3 The Case for TCP
This is one of the most hotly debated issues in the wireless
arena. Here are some arguments against it:
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- It is generally recognized that TCP does not perform well in
the presence of significant levels of non-congestion loss.
TCP detractors argue that the wireless medium is one such
case, and that it is hard enough to fix TCP. They argue that
it is easier to start from scratch.
- TCP has too much header overhead.
- By the time the mechanisms are in place to fix it, TCP is
very heavy, and ill-suited for use by lightweight, portable
devices.
and here are some in support of TCP:
- It is preferrable to continue using the same protocol that
the rest of the Internet uses for compatibility reasons. Any
extensions specific to the wireless link may be negotiated.
- Legacy mechanisms may be reused (for example congestion
control)
- Link-layer FEC and ARQ can reduce the BER such that any
losses TCP does see are, in fact, caused by congestion (or a
sustained interruption of link connectivity). Modern W-WAN
technologies do this (CDPD, US-TDMA, CDMA, GSM), thus
improving TCP throughput.
- Handoffs among different technologies are made possible by
Mobile IP [RFC2002], but only if the same protocols, namely
TCP/IP, are used throughout.
- Given TCP's wealth of research and experience, alternative
protocols are relatively immature, and the full implications
of their widespread deployment not clearly understood.
Overall, we feel that TCP is fixable. Mechanisms to do so are
included in the next sections.
4 Candidate Optimizations
There is a large volume of work on the subject of optimizing TCP
for operation over wireless media. Even though satellite networks
generally fall in the LFN regime, our current LTN focus has much
to benefit from it. For example, the work of the
TCP-over-Satellite working group of the IETF has been extremely
helpful in preparing this section [AGS98, ADGGHOSSTT98].
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4.1 TCP: Current Mechanisms
A TCP sender adapts its use of bandwidth based on feedback from
the receiver. The high latency characteristic of LTNs implies that
TCP's adaptation is correspondingly slower than on networks with
shorter delays. Delayed ACKs and small MTUs may slow adaptation
even further.
4.1.1 Slow Start and Congestion Avoidance
Slow Start and Congestion Avoidance [RFC2001, updated in TCPCONG]
are the basis for TCP's successful deployment throughout the
internet. However there are two reasons why the wireless medium
adversely affects them:
- Slow start is invoked whenever a loss is detected, assuming
the network is congested. This is why it is important to
minimize the losses caused by corruption, leaving only those
that TCP expects.
- The sender increases its window based on the number of ACKs
received. This rate, of course, is dependent on the RTT
(round-trip-time) between sender and receiver, which implies
long delays in high latency links like LTNs.
- During slow start, the sender increases its window in units
of segments. This is why it is important to use an
appropriately large MTU which, in turn, requires reliable
link layers.
4.1.2 Fast Retransmit and Fast Recovery
Fast retransmit [RFC2001, updated in TCPCONG] allows the receiver
to inform the sender (by sending several duplicate ACKs) of a lost
segment. The sender retransmits what it considers to be this lost
segment without waiting for the full timeout. This saves time.
After a fast retransmit, a sender invokes the fast recovery
[RFC2001] algorithm, whereby it invokes congestion avoidance, but
not slow start. This also saves time.
With LTN links, efficient recovery from multiple losses within a
single window is more difficult to achieve, and waiting for three
duplicate ACKs to arrive postpones retransmission noticeably.
Recommendation: Implement at this time. This is a
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widely-implemented optimization and is currently at Proposed
Standard level. [AGS98] recommends implementation of Fast
Retransmit/Fast Recovery in satellite environments.
4.2 Connection Setup with T/TCP [RFC1397, RFC1644]
TCP engages in a "three-way handshake" whenever a new connection
is set up. Data transfer is only possible after this phase has
completed successfuly. T/TCP allows data to be exchanged in
parallel with the connection set up, saving valuable time for
short transactions on long-latency networks.
Recommendation: T/TCP is not recommended, for these reasons:
- It is an Experimental RFC, and has not been advanced.
- It is not widely deployed, and it has to be deployed at both
ends of a connection.
- Security concerns have been raised that T/TCP is more vulnerable
to address-spoofing attacks than TCP itself.
- At least some of the benefits of T/TCP (eliminating three-way
handshake on subsequent query-response transactions, for
instance) are also available with persistent connections on
HTTP/1.1, which is more widely deployed.
[ADGGHOSSTT98] does not have a recommendation on T/TCP in
satellite environments.
4.3 Slow Start Proposals
Because slow start dominates the network response seen by
interactive users at the beginning of a TCP connection, a number
of proposals have been made to modify or or eliminate slow start
in long latency environments.
Stability of the internet is paramount, so these proposals must
demonstrate that they will not adversely affect internet
congestion levels in significant ways. The needs of the many
outweigh the needs of the few.
4.3.1 Larger Initial Window
Full slow start, with an initial window of one segment, is a
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time-consuming bandwidth adaptation procedure over LTNs. Recent
proposals suggest starting off with an initial window larger than
one segment [RFC2414, AHO98].
In simulations with an increased initial window of three packets
[RFC2415], this proposal does not contribute significantly to
packet drop rates, and it has the added benefit of improving
initial response times when the peer device delays
acknowledgements during slow start (see next proposal).
[RFC2416] addresses situations where the initial window exceeds
the number of buffers available to TCP and indicates that this
situation is no different from the case where the congestion
window grows beyond the number of buffers available.
We expect the IETF tcp-impl working group to recommend allowing an
initial window of at least two segments, and perhaps as many as
four, in the near future, in environments where this significantly
improves performance (LFNs and LTNs).
Recommendation: Implement this on devices now. The research on
this optimization indicates that 3 segments is a safe initial
setting, and is centering on choosing between 2, 3, and 4. For
now, use 3 [RFC2415], which at least allows clients running
query-response applications to get an initial ACK from unmodified
servers without waiting for a delayed ACK timeout of 200-500
milliseconds, and saves two round-trips.
Much of the benefit of this optimization is also available (after
the first request-response exchange) when clients and servers both
implement HTTP/1.1. This optimization works with older servers
that implement only HTTP/1.0.
4.3.2 Handling Acknowledgments During Slow Start
The sender increases its window based on the flow of ACKs coming
back from the receiver. Particularly during slow start, this flow
is very important. A couple of the proposals that have been
studied are (1) ACK counting and (2) ACK-every-segment.
4.3.2.1 ACK Counting
The main idea behing ACK counting is:
- Make each ACK count to its fullest by growing the window
based on the data being acknowledged (byte counting) instead
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of the number of ACKs (ACK counting). This has been shown to
cause bursts which lead to congestion. [Allman98] shows that
Limited Byte Counting (LBC), in which the window growth is
limited to 2 segments, does not lead to burstiness, and
offers some performance gains.
Recommendation: Unlimited byte counting is not recommended. Van
Jacobson cautions against byte counting [TCPSATMIN] because it
leads to burstiness, and recommends ACK spacing [ACKSPACING]
instead.
ACK spacing requires ACKs to consistently pass through a single
ACK-spacing router. This requirement works well for W-WAN
environments if the ACK-spacing router is also the base station.
Limited byte counting warrants further investigation before we can
recommend this proposal, but it shows promise.
4.3.2.2 ACK-every-segment
The main idea behind ACK-every-segment is:
- Keep a constant stream of ACKs coming back by turning off
delayed ACKs [RFC1122] during slow start. ACK-every-segment
must be limited to slow start, in order to avoid penalizing
asymmetric-bandwidth configurations.
Recommendation: Implement this on devices but continue
investigating. Even though simulations confirm its promise (it
allows receivers to receive the second segment from unmodified
senders without waiting for a delayed ACK timeout of 200-500
milliseconds), for this technique to be practical the receiver
must acknowledge every segment only when the sender is in slow
start. Continuing to do so when the sender is in congestion
avoidance may have adverse effects on the mobile device's battery
consumption and on traffic in the network.
This violates a SHOULD in [TCPCONG]: delayed acknowledgements
SHOULD be used by a TCP receiver.
"Disabling Delayed ACKs During Slow Start" is technically
unimplementable, as the receiver has no way to know when the
sender crosses ssthresh (the "slow start threshold") and begins
using the congestion avoidance algorithm. If receivers follow
recommendations for increased initial windows, disabling delayed
ACKs during an increased initial window would open the TCP window
more rapidly without doubling ACK traffic in general.
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The conservative recommendation is to ACK only the first segment
on a new connection with no delay. Even this conservative
recommendation saves one delayed ACK timeout at the receiver,
which, in typical WWW applications, saves one delayed ACK timeout
in each direction.
Again, much of the benefit of this optimization is also available
after the first request-response exchange when clients and servers
both implement HTTP/1.1. This optimization works with older
servers that implement only HTTP/1.0.
4.3.3 Terminating Slow Start
New mechanisms [ADGGHOSSTT98] are being proposed to improve TCP's
adaptive properties such that the available bandwidth is better
utilized while reducing the possibility of congesting the
network. The latter leads to the closing of the congestion window
to 1 segment, and the subsequent slow start phase.
Theoretically, an optimum value for slow-start threshold
(ssthresh) allows connection bandwidth utilization to ramp up as
aggressively as possible without "overshoot" (using so much
bandwidth that packets are lost and congestion avoidance
procedures are invoked).
Recommendation: Estimating the slow start threshold is not
recommended. Although this would be helpful if we knew how to do
it, rough consensus on the tcp-impl and tcp-sat mailing lists is
that in non-trivial operational networks there is no reliable
method to probe during TCP startup and estimate the bandwidth
available.
4.4 ACK Spacing
During slow start, the sender responds to the incoming ACK stream
by transmitting two new segments for each ACK received. This
results in data being sent at twice the speed at which it can be
processed by the network. Accordingly, queues will form, and due
to insufficient buffering at the bottleneck router, packets may
get dropped before the link's capacity is full. Spacing out the
ACKs effectively controls the rate at which the sender will
transmit into the network, and may result in little or no queueing
at the bottleneck router [ACKSPACING].
Recommendation: No recommendation at this time. Continue
monitoring research in this area.
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4.5 Delayed Duplicate Acknowlegements
As was mentioned above, link-layer retransmissions may decrease
the BER enough that congestion accounts for most of packet losses;
still, nothing can be done about interruptions due to handoffs,
moving beyond wireless coverage, etc. In this scenario, it is
imperative to prevent interaction between link-layer
retransmission and TCP retransmission as these layers duplicate
each other's efforts. In such an environment it may make sense to
delay TCP's efforts so as to give the link-layer a chance to
recover [MV97]. It is preferrable to allow a local mechanism to
resolve a local problem, instead of invoking TCP's end-to-end
mechanism and incurring the associated costs, both in terms of
wasted bandwidth and in terms of its effect on TCP's window.
Recommendation: Delaying duplicate acknowledgements may be useful
in specific network topologies, but a general recommendation
requires further research and experience. Currently, it is not
well understood how long the receiver should delay the duplicate
acknowledgments.
4.6 Selective Acknowledgements [RFC2018]
SACK may not be useful in many LTNs, according to Section 1.1 of
[TCPHP]. In particular, SACK is more useful in the LFN regime,
especially if large windows are being used, because there is a
considerable probability of multiple segment losses per window. In
the LTN regime, TCP windows are much smaller, and burst errors
must be much longer in duration in order to damage multiple
segments.
Accordingly, the complexity of SACK may not be justifiable, unless
there is a high probability of burst errors and congestion on the
wireless link. A desire for compatibility with TCP recommendations
for non-LTN environments may dictate LTN support for SACK anyway.
[AGS98] recommends use of SACK with Large TCP Windows in satellite
environments, and notes that this implies support for PAWS
(Protection Against Wrapped Sequence space) and RTTM (Round Trip
Time Measurement) as well.
Berkeley's SNOOP protocol research [SNOOP] indicates that SACK
does improve throughput for SNOOP when multiple segments are lost
per window [BPSK96]. SACK allows SNOOP to recover from
multi-segment losses in one round-trip. In this case, the mobile
device needs to implement some form of selective
acknowledgements. If SACK is not used, recovery from
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multi-segment losses takes so long that TCP enters congestion
avoidance anyway.
Recommendation: Implement SACK now for compatibility with other
TCPs and improved performance with SNOOP.
4.7 Detecting Corruption Loss
4.7.1 Without Explicit Notification
In the absence of explicit notification from the network, some
researchers have suggested statistical methods for congestion
avoidance [Jain89, WC91, VEGAS]. A natural extension of these
heuristics would enable a sender to distinguish between losses
caused by congestion and other causes. The research results on
the reliability of sender-based heuristics is unfavorable [BV97,
BV98]. [BV98a] reports better results in constrained environments
using packet inter-arrival times measured at the receiver, but
highly-variable delay - of the type encountered in wireless
environments during intercell handoff - confounds these
heuristics.
Recommendation: No recommendation at this time - continue to
monitor research results.
4.7.2 With Explicit Notifications
With explicit notification from the network it is possible to
determine when a loss is due to congestion. Several proposals
along these lines include:
- Explicit Loss Notification (ELN) [BPSK96]
- Explicit Bad State Notification (EBSN) [BBKVP96]
- Explicit Loss Notification to the Receiver (ELNR), and
Explicit Delayed Dupack Activation Notification (EDDAN)
(notifications to mobile receiver) [MV97]
- Explicit Congestion Notification (ECN) [ECN]
Of these proposals, Explicit Congestion Notification (ECN)
seems closest to deployment on the Internet, and will provide
some benefit for TCP connections on long thin networks (as well
as for all other TCP connections).
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Recommendation: No recommendation at this time. Schemes like ELNR
and EDDAN [MV97], in which the only systems that need to be
modified are the base station and the mobile device are slated for
adoption pending further research. However, the requirement that
the base station be able to examine the TCP headers flying through
it raises the previously stated issues with respect to
IPSEC-encrypted packets.
ECN uses the TOS byte in the IP header to carry congestion
information (ECN-capable and Congestion-encountered). This byte
is not encrypted in IPSEC, so ECN can be used on TCP connections
that are encrypted using IPSEC.
Recommendation: Implement ECN when (and if) the IETF finalizes the
specification.
Note: Absence of packets marked with ECN should not be interpreted
by ECN-capable TCP connections as a green light for aggressive
retransmissions. On the contrary, during periods of extreme
network congestion routers may drop packets marked with explicit
notification because their buffers are exhausted - exactly the
wrong time for a host to begin retransmitting aggressively.
4.8 Active Queue Management
As has been pointed out above, TCP responds to congestion by
closing down the window and invoking slow start. Long-delay
networks take a particularly long time to recover from this
condition. Accordingly, it is imperative to avoid congestion in
LTNs. To remedy this, active queue management techniques have been
proposed as enhancements to routers throughout the Internet [RED].
The primary motivation for deployment of these mechanisms is to
prevent "congestion collapse" (a severe degradation in service) by
controlling the average queue size at the routers. As the average
queue length grows, Random Early Detection [RED] increases the
possibility of dropping packets.
The benefits are:
- Reduce packet drops in routers. By dropping a few packets
before severe congestion sets in, RED avoids dropping bursts
of packets.
- Provide lower delays. This follows from the smaller queue
sizes, and is particularly important for interactive
applications, for which the inherent delays of wireless links
already push the user experience to the limits of the
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non-acceptable.
- Avoid lock-outs. Packets from over-agressive flows can get
dropped with the same probability as other packets.
Active Queue Management has two components: (1) routers detect
congestion before exhausting their resources, and (2) they provide
some form of congestion indication. Dropping packets via RED is
only one example of the latter. Another way to indicate congestion
is to use ECN [ECN] as discussed above under "Detecting Corruption
Loss: With Explicit Notifications."
Recommendation: RED is currently being deployed in the internet,
and LTNs should follow suit. ECN deployment should complement RED's
when the IETF finalizes the specification.
4.9 Scheduling Algorithms
Active queue management helps control the length of the queues.
Additionally, a general solution requires replacing FIFO with
other scheduling algorithms that improve:
1. Fairness (by policing how different packet streams utilize
the available bandwidth), and
2. Throughput (by improving the transmitter's radio channel
utilization).
For example, fairness is necessary for interactive applications
(like telnet or web browsing) to coexist with bulk transfer
sessions. Proposals here include:
- Fair Queueing (FQ) [Demers90]
- Class-based Queueing (CBQ) [Floyd95]
Even if they are only implemented over the wireless link portion
of the communication path, these proposals are attractive in
wireless LTN environments, because new connections for interactive
applications can have difficulty starting when a bulk TCP transfer
has already stabilized using all available bandwidth.
In our base architecture described above, the mobile device
typically communicates directly with only one wireless peer at a
given time: the base station. In some W-WANs, it is possible to
directly address other mobiles within the same cell. Direct
communication with each such wireless peer may traverse a
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spatially distinct path, each of which may exhibit statistically
independent radio link characteristics. Channel State Dependent
Packet Scheduling (CSDP) [BBKT96] tracks the state of the various
radio links (as defined by the target devices), and gives
preferential treatment to packets destined for radio links in a
"good" state. This avoids attempting to transmit to (and expect
acknowledgements from) a peer on a "bad" radio link, thus
improving throughput.
A further refinement of this idea suggests that both fairness and
throughput can be improved by combining a wireless-enhanced CBQ
with CSDP [FSS98].
Recommendation: No recommendation at this time, pending further
study.
4.10 Split TCP and Performance-Enhancing Proxies (PEPs)
Given the dramatic differences between the wired and the wireless
links, a very common approach is to provide some impedance
matching where the two different technologies meet: at the base
station.
The idea is to replace an end-to-end TCP connection with two
clearly distinct connections: one across the wireless link, the
other across its wireline counterpart. Each of the two resulting
TCP sessions operates under very different networking
characteristics, and may adopt the policies best suited to its
particular medium. For example, in a specific LTN topology it may
be desirable to modify TCP Fast Retransmission to resend after the
first duplicate ack and Fast Recovery not to shrink TCP cnwd if
the LTN link has an extremely long RTT, is known to not reorder
packets, and is not subject to congestion. Moreover, on a
long-delay link or on a link with a relatively high
bandwidth-delay product it may be desirable to "slow-start" with a
relatively large initial window, even larger than four segments.
While these kinds of TCP modifications can be negotiated to be
employed over the LTN link, they would not be deployed end-to-end
over the global Internet. In LTN topologies where the underlying
link characteristics are known, a great number of similar type of
performance enhancements can be employed without endangering
operations over the global Internet.
Split-TCP proposals include schemes like I-TCP [ITCP] which
achieves performance improvements by abandoning end-to-end
semantics. [MTCP] alleviates this problem somewhat, but does not
entirely solve it. The Mowgli architecture [MOWGLI] proposes a
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split approach with support for various enhancements at all the
protocol layers, not only at the transport layer. Mowgli provides
an option to replace the TCP/IP core protocols on the LTN link
with a custom protocol that is tuned for LTN links [KRLKA97].
Also with this option, Mowgli preserves the socket semantics on
the mobile device so that legacy applications can be run
unmodified.
With LTN links, significant improvements can be achieved at the
application layer by introducing application level proxies. The
Mowgli system provides full support for adding application level
agent-proxy pairs between the client and the server, the agent on
the mobile device and the proxy on the LTN-edge device. Such a
pair may be either explicit or fully transparent to the
applications. Good examples of enhancements achieved with
application level proxies include Mowgli WWW [LAKLR95], [LHKR96]
and WebExpress [HL96],[CTCSM97].
Berkeley's SNOOP protocol [SNOOP] is a hybrid scheme mixing
link-layer reliability mechanisms with the split connection
approach. It is an improvement over I-TCP in that end-to-end
semantics are retained as well as in terms of performance. SNOOP
does two things:
1. Locally (on the wireless link) retransmit lost packets,
instead of allowing TCP to do so end-to-end.
2. Suppress the duplicate acks on their way from the receiver
back to the sender, thus avoiding fast retransmit and
congestion avoidance at the latter.
WTCP [WTCP] is similar to SNOOP in that it preserves end-to-end
semantics. In WTCP, the base station uses a complex scheme to
hide the time it spends moving packets across the wireless link
(this typically includes retransmissions due to error recovery,
but may also include time spent dealing with congestion). In order
to work effectively, it assumes that the TCP endpoints implement
the Timestamps option in RFC 1323 [TCPHP]. Unfortunately, support
for RFC 1323 in TCP implementations is not yet widespread. Beyond
this, WTCP requires changes only at the base station.
All of these schemes require the base station to examine and
operate on the traffic between the portable wireless device and
the TCP server on the wired Internet. None of these work if the IP
traffic is encrypted, unless, of course, the base station shares
the security association between the mobile device and its
end-to-end peer. They also require that both the data and the
corresponding ACKs traverse the same base station. Furthermore,
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if the base station retransmits packets at the transport layer
across the wireless link, this may duplicate efforts by the
link-layer. SNOOP has been described by its designers as a
TCP-aware link-layer. This is the right approach: the link and
network layers can be much more aware of each other than
traditional OSI layering suggests.
Encryption of IP packets via IPSEC's ESP header (in either
transport or tunnel mode) renders the TCP header and payload
unintelligible to the base station. This precludes SNOOP from
working, because it needs to examine the TCP headers in both
directions. Possible solutions involve:
- making the SNOOPing base station a party to the security
association between the client and the server
- IPSEC tunneling mode, terminated at the SNOOPing base station
However, these techniques require that users trust base stations.
Users valuing both privacy and performance should use SSL or SOCKS
for end-to-end security.
Recommendation: Implement SNOOP on base stations now. Research
results are encouraging, and it is an "invisible" optimization in
that neither the client nor the server needs to change, only the
base station (for basic SNOOP without SACK).
In some proposals, in addition to a PEP mechanism at the base
station, custom protocols are used on the wireless link (for
example, [WAP], [YB94] or [MOWGLI]).
Even if the gains from using non-TCP protocols are moderate or
better, the wealth of research on optimizing TCP for wireless, and
compatibility with the Internet are compelling reasons to adopt
TCP on the wireless link (enhanced as suggested in section 5
below).
4.11 Header Compression Alternatives
Because Long Thin Networks are bandwidth-constrained, compressing
every byte out of over-the-air segments is worth while.
Mechanisms for TCP and IP header compression defined in [RFC1144,
IPHC, IPHC-PPP] provide the following benefits:
- Improve interactive response time
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- Allow using small packets for bulk data with good line
efficiency
- Allow using small packets for delay sensitive low data-rate
traffic
- Decrease header overhead (for the smallest MTU of 512 the
header overhead of TCP over IP can decrease from 11.7 to less
than 1 per cent.
- Reduce packet loss rate over lossy links.
Van Jacobson (VJ) header compression [RFC1144] describes a
Proposed Standard for TCP Header compression that is widely
deployed. It uses TCP timeouts to detect a loss of
synchronization between the compressor and decompressor. [IPHC]
includes an explicit request for retransmission of an uncompressed
packet to allow resynchronization without waiting for a TCP
timeout (and executing congestion avoidance procedures).
Recommendation: Implement VJ header compression on devices now.
It is a widely deployed Proposed Standard. However, it should
only be enabled when operating over reliable LTNs, because even a
single bit error would most probably result in a full TCP window
being dropped, followed by a costly recovery via slow-start.
Implement [IPHC] when the Internet-Draft becomes stable.
4.12 IP Payload Compression
Compression of IP payloads is also desirable. "IP Payload
Compression Protocol (IPComp)" [IPPCP] defines a framework where
common compression algorithms can be applied to arbitrary IP
segment payloads. IP payload compression is something of a niche
optimization. It is necessary because IP-level security converts
IP payloads to random bitstreams, defeating commonly-deployed
link-layer compression mechanisms which are faced with payloads
that have no redundant "information" that can be more compactly
represented.
However, many IP payloads are already compressed (images, audio,
video, "zipped" files being FTPed), or are already encrypted above
the IP layer (SSL/TLS, etc.). These payloads will not "compress"
further, limiting the benefit of this optimization.
HTTP-NG is considering supporting compression of resources at the
HTTP level, which would provide equivalent benefits for common
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compressible MIME types like text/html. This will reduce the need
for IPComp. If IPComp is deployed more rapidly than HTTP-NG,
IPComp compression of HTML and MIME headers would be beneficial.
In general, application-level compression can often outperform
IPComp, because of the opportunity to use compression dictionaries
based on knowledge of the specific data being compressed.
Recommendation: Track IPComp and HTTP-NG standardization and
deployment for now.
4.13 TCP Control Block Interdependence [Touch97]
TCP maintains per-connection information such as connection state,
current round-trip time, congestion control or maximum segment
size. Sharing information between two consecutive connections or
when creating a new connection while the first is still active to
the same host may improve performance of the latter connection.
The principle could easily be extended to LAN coverage rather than
limiting itself to hosts. [Touch97] describes cache update for
both cases.
Users of W-WAN devices frequently request connections to the same
servers or set of servers. For example, in order to read their
email or to initiate connections to other servers, the devices may
be configured to always use the same email server or WWW proxy.
The main advantage of this proposal is that it relieves the
application of the burden of optimizing the transport layer. In
order to improve the performance of TCP connections, this
mechanism only requires changes at the wireless device.
In general, this scheme should improve the dynamism of connection
setup without increasing the cost of the implementation.
Recommendation: Recommended for implementation. Monitor research
on this.
5 Summary of Recommended Optimizations
The table below summarizes our recommendations with regards to the
main proposals mentioned above.
The first column, "Stability of the Proposal," refers to the
maturity of the mechanism in question. Some proposals are
being pursued within the IETF in a somewhat open fashion. An
IETF proposal is either an Internet Drafts (I-D) or a Request
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for Comments (RFC). The former is a preliminary version. There
are several types of RFCs. A Draft Standards (DS) is standards
track, and carries more weight than a Proposed Standard (PS),
which may still undergo revisions. Informational or Experimental
RFCs do not specify a standard. Other proposals are isolated
efforts with little or no public review, and little chance of
garnering industry backing.
"Implemented at" indicates which participant in a TCP session
must be modified to implement the proposal. Legacy servers
typically cannot be modified, so this column indicates whether
implementation happens at either or both of the two nodes under
some control: mobile device and base station. The symbols used
are: WS (wireless sender, that is, the mobile device's TCP send
operation must be modified), WR (wireless receiver, that is,
the mobile device's TCP receive operation must be modified), WD
(wireless device, that is, modifications at the mobile device are
not specific to either TCP send or receive), BS (base station)
and NI (network infrastructure).
The "Recommendation" column captures our suggestions. Some
mechanisms are endorsed for immediate adoption, others need more
evidence and research, and others are not recommended.
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Name Stability of Implemented Recommendation
the Proposal at
==================== ============= =========== =================
Increased Initial RFC 2414 (EXP) WS Yes
Congestion Window (initial_window=3)
Disable delayed ACKs NA WR When stable
during slow start
Byte counting NA WS No
instead of ACK
counting
TCP Header RFC 1144 (PS) WD Yes
compression for PPP BS (see 4.11)
IP Payload IETF I-D WD When stable
Compression (Approved as (simultaneously
(IPComp) PS) needed on Server)
IP Header IETF I-D WD When stable
Compression BS (For Mobile IP only)
(includes IP-IP) for
PPP
SNOOP plus SACK In limited use BS Yes
WD (for SACK)
Fast retransmit/fast RFC 2001 (PS) WD Yes (should be
recovery there already)
Transaction/TCP RFC 1644 WD No
(Experimental) (simultaneously
needed on Server)
Estimating Slow NA WS No
Start Threshold
(ssthresh)
Delayed Duplicate Not stable WR When stable
Acknowledgements BS (for
notifications)
Class-based Queuing NA WD When stable
on End Systems
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Explicit Congestion IETF I-D WD When stable
Notification NI
TCP Control Block RFC 2140 WD Yes
Interdependence (Informational) (Track research)
Of all the optimizations in the table above, only SNOOP plus SACK
and Delayed duplicate acknowledgements are currently being
proposed only for wireless networks. The others are being
considered even for non-wireless applications. Their more general
applicability attracts more attention and analysis from the
research community.
Of the above mechanisms, only Header Compression (for IP and TCP)
and "SNOOP plus SACK" cease to work in the presence of IPSec.
6 Conclusion
In view of the unpredictable and problematic nature of long thin
networks, arriving at an optimized transport is a daunting task. We
have reviewed the existing proposals along with future research
items. Based on this overview, we also recommend mechanisms for
implementation in long thin networks (LTNs).
7 Acknowledgements
The authors are deeply indebted to the IETF tcpsat and tcpimpl
working groups, and to Prof. Nitin Vaidya (Texas A&M) whose many
insightful comments have proved invaluable in our attempt at
improving this note. The following individuals have also provided
valuable feedback: Mark Allman (NASA), Raphi Rom (Technion/Sun),
Charlie Perkins (Sun), Peter Stark (Ericsson).
8 References
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Paths with Insufficient Buffering," September 1998. Internet-Draft
draft-rfced-info-partridge-01.txt (work in progress).
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Griner, John Heidemann, Shawn Osterman, Keith Scott, Jeffrey
Semke, Joe Touch, Diepchi Tran. Ongoing TCP Research Related to
Satellites, August 1998. Internet-Draft
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April 1993.
[HL96] Hausel, B., Lindquist, D., "WebExpress: A System for
Optimizing Web Browsing in a Wireless Environment," in Proc.
MobiCom'96, Rye, New York, USA, November 1996.
[IPPCP] A. Shacham, R. Monsour, R. Pereira, M. Thomas, IP Payload
Compression Protocol (IPComp), May 1998. Internet-Draft
draft-ietf-ippcp-protocol-06.txt (work in progress).
[IPHC] Mikael Degermark, Bjorn Nordgren,Stephen Pink. IP Header
Compression, June 1998. Internet-Draft
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INTERNET DRAFT Long Thin Networks November 1998
draft-degermark-ipv6-hc-06.txt (work in progress).
[IPHC-PPP] Mathias Engan, S. Casner, C. Bormann. IP Header
Compression over PPP, December 1997. Internet-Draft
draft-engan-ip-compress-02.txt (work in progress).
[ITCP] Bakre, A., Badrinath, B.R., "Handoff and Systems Support
for Indirect TCP/IP. In Proceedings of the Second USENIX Symposium
on Mobile and Location-Independent Computing, Ann Arbor, Michigan,
April 10-11, 1995.
[Jain89] Jain, R., "A Delay-Based Approach for Congestion
Avoidance in Interconnected Heterogeneous Computer Networks,"
Digital Equipment Corporation, Technical Report DEC-TR-566, April
1989.
[KRLKA97] Kojo, M., Raatikainen, K., Liljeberg, M., Kiiskinen,
J., Alanko, T., "An Efficient Transport Service for Slow Wireless
Telephone Links," in IEEE Journal on Selected Areas of
Communication, volume 15, number 7, September 1997.
[LAKLR95] Liljeberg, M., Alanko, T., Kojo, M., Laamanen, H.,
Raatikainen, K., "Optimizing World-Wide Web for Weakly-Connected
Mobile Workstations: An Indirect Approach," in Proc. 2nd Int.
Workshop on Services in Distributed and Networked Environments,
Whistler, Canada, pp. 132-139, June 1995.
[LHKR96] Liljeberg, M., Helin, H., Kojo, M., Raatikainen, K.,
"Mowgli WWW Software: Improved Usability of WWW in Mobile WAN
Environments," in Proc. IEEE Global Internet 1996 Conference,
London, UK, November 1996.
[LS98] Lettieri, P., Srivastava, M., "Adaptive Frame Length
Control for Improving Wireless Link Throughput, Range, and Energy
Efficiency," Proc. IEEE INFOCOM'98, April 1998.
[MNCP] Piscitello, D., Phifer, L., Wang, Y., Hovey, R., Mobile
Network Computing Protocol (MNCP), August 1997. Internet-Draft
draft-piscitello-mncp-00.txt (work in progress).
[MOWGLI] Kojo, M., Raatikainen, K., Alanko, T., "Connecting Mobile
Workstations to the Internet over a Digital Cellular Telephone
Network," in Proc. Workshop on Mobile and Wireless Information
Systems (MOBIDATA), Rutgers University, NJ, November 1994.
Available at: http://www.cs.Helsinki.FI/research/mowgli/. Revised
version published in Mobile Computing, pp. 253-270, Kluwer, 1996.
[MSMO97] Mathis, M., Semke, J., Mahdavi, J., Ott, T., "The
Montenegro,Dawkins Expires May 23, 1999 [Page 32]
INTERNET DRAFT Long Thin Networks November 1998
Macroscopic Behavior of the TCP Congestion Avoidance Algorithm,"
in Computer Communications Review, a publication of ACM SIGCOMM,
volume 27, number 3, July 1997.
[MTCP] Brown, K. Singh, S., "A Network Architecture for Mobile
Computing," Proc. IEEE INFOCOM'96, pp. 1388-1396, March 1996.
Available at
ftp://ftp.ece.orst.edu/pub/singh/papers/transport.ps.gz.
[MV97] Mehta, M., Vaidya, N., "Delayed
Duplicate-Acknowledgements: A Proposal to Improve Performance of
TCP on Wireless Links," Texas A&M University, December 24, 1997.
Available at http://www.cs.tamu.edu/faculty/vaidya/mobile.html
[NETBLT] John C. White, NETBLT (Network Block Transfer Protocol),
draft-white-protocol-stack-00.txt, April 1997 - work in progress.
[RED] Braden, B. Clark, D., Crowcroft, J., Davie, B., Deering, S.,
Estrin, D., Floyd, S., Jacobson, V., Minshall, G., Partridge, C.,
Peterson, L., Ramakrishnan, K.K., Shenker, S., Wroclawski, J.,
Zhang, L., "Recommendations on Queue Management and Congestion
Avoidance in the Internet," RFC 2309, April 1998.
[RFC908] Velten, D., Hinden, R., Sax, J., "Reliable Data
Protocol", RFC 908, July 1984.
[RFC1030] Lambert, M., "On Testing the NETBLT Protocol over Divers
Networks", RFC 1030, November 1987.
[RFC1122] Braden, R., Requirements for Internet Hosts --
Communication Layers, October 1989.
[RFC1144] Jacobson, V., Compressing TCP/IP Headers for Low-Speed
Serial Links, RFC 1144, February 1990.
[RFC1151] Partridge, C., Hinden, R., Version 2 of the Reliable
Data Protocol (RDP), RFC 1151, April 1990.
[RFC1191] Jeff Mogul and Steve Deering. Path MTU Discovery,
November 1990. RFC 1191.
[RFC1397] Braden, R., Extending TCP for Transactions -- Concepts,
November 1992.
[RFC1644] Braden, R., T/TCP -- TCP Extensions for Transactions
Functional Specification, July 1994.
[RFC1661] Simpson, W., ed., "The Point-To-Point Protocol (PPP)",
Montenegro,Dawkins Expires May 23, 1999 [Page 33]
INTERNET DRAFT Long Thin Networks November 1998
RFC 1661, July 1994.
[RFC1986] Polites, W., Wollman, W., Woo, D., Langan, R.,
"Experiments with a Simple File Transfer Protocol for Radio Links
using Enhanced Trivial File Transfer Protocol (ETFTP)", RFC 1986,
August 1996.
[RFC2001] W. Richard Stevens. TCP Slow Start, Congestion
Avoidance, Fast Retransmit, and Fast Recovery Algorithms, January
1997. RFC 2001.
[RFC2002] Perkins, C., Editor, "IP Mobility Support", RFC 2002,
October 1996.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and Romanow, A.,
"TCP Selective Acknowledgment Options", October, 1996.
[RFC2188] Banan, M., Taylor, M., Cheng, J., "AT&T/Neda's Efficient
Short Remote Operations (ESRO) Protocol Specification Version
1.2", RFC 2188, September 1997.
[RFC2414] Mark Allman, Sally Floyd, Craig Partridge. Increasing
TCP's Initial Window, September 1998. RFC 2414.
[RFC2415] Poduri, K., Nichols, K. Simulation Studies of Increased
Initial TCP Window Size, September 1998. RFC 2415.
[RFC2416] Tim Shepard and Craig Partridge. When TCP Starts Up With
Four Packets Into Only Three Buffers, September 1998. RFC 2416.
[SNOOP] Balakrishnan, H., Seshan, S., Amir, E., Katz, R.,
"Improving TCP/IP Performance over Wireless Networks," Proc. 1st
ACM Conf. on Mobile Computing and Networking (Mobicom), Berkeley,
CA, November 1995.
[TCPCONG] W. Richard Stevens, Mark Allman, Vern Paxson. TCP
Congestion Control, November 1998. Internet-Draft
draft-ietf-tcpimpl-cong-control-01.txt (work in progress).
[TCPHP] Van Jacobson, Robert Braden, and David Borman. TCP
Extensions for High Performance, May 1992. RFC 1323.
[TCPSATMIN] TCPSAT Minutes, August, 1997. Available at:
http://tcpsat.lerc.nasa.gov/tcpsat/meetings/munich-minutes.txt.
[Touch97] Touch, T., "TCP Control Block Interdependence," RFC
2140, April 1997.
Montenegro,Dawkins Expires May 23, 1999 [Page 34]
INTERNET DRAFT Long Thin Networks November 1998
[VEGAS] Brakmo, L., O'Malley, S., "TCP Vegas, New Techniques for
Congestion Detection and Avoidance," SIGCOMM'94, London, pp 24-35,
October 1994.
[VMTP] Cheriton, D., "VMTP: Versatile Message Transaction
Protocol", RFC 1045, February 1988.
[WAP] Wireless Application Protocol Forum.
http://www.wapforum.org/
[WC91] Wang, Z., Crowcroft, J., "A New Congestion Control Scheme:
Slow Start and Search," ACM Computer Communication Review, vol 21,
pp 32-43, January 1991.
[WTCP] Ratnam, K., Matta, I., "WTCP: An Efficient Transmission
Control Protocol for Networks with Wireless Links," Technical
Report NU-CCS-97-11, Northeastern University, July 1997. Available
at: http://www.ece.neu.edu/personal/karu/papers/WTCP-NU.ps.gz
[YB94] Yavatkar, R., Bhagawat, N., "Improving End-to-End
Performance of TCP over Mobile Internetworks," Proc. Workshop on
Mobile Computing Systems and Applications, IEEE Computer Society
Press, Los Alamitos, California, 1994.
Authors' addresses
Questions about this document may be directed at:
Montenegro,Dawkins Expires May 23, 1999 [Page 35]
INTERNET DRAFT Long Thin Networks November 1998
Gabriel E. Montenegro
Sun Labs Networking and Security Group
Sun Microsystems, Inc.
901 San Antonio Road
Mailstop UMPK 15-214
Mountain View, California 94303
Voice: +1-650-786-6288
Fax: +1-650-786-6445
E-Mail: gab@sun.com
Spencer Dawkins
Nortel Networks
P.O. Box 833805
Richardson, Texas 75083-3805
Voice: +1-972-684-4827
Fax: +1-972-685-3292
E-Mail: sdawkins@nortel.com
Markku Kojo
University of Helsinki/Department of Computer Science
P.O. Box 26 (Teollisuuskatu 23)
FIN-00014 HELSINKI
Finland
Voice: +358-9-7084-4179
Fax: +358-9-7084-4441
E-Mail: kojo@cs.helsinki.fi
Vincent Magret
Corporate Research Center
Alcatel Network Systems, Inc
1201 Campbell
Mail stop 446-310
Richardson Texas 75081 USA
M/S 446-310
Voice: +1-972-996-2625
Fax: +1-972-996-5902
E-mail: vincent.magret@aud.alcatel.com
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