Internet DRAFT - draft-montenegro-pilc-ltn
draft-montenegro-pilc-ltn
Internet Engineering Task Force G. Montenegro
INTERNET DRAFT Sun Microsystems, Inc.
S. Dawkins
Nortel Networks
M. Kojo
University of Helsinki
V. Magret
Alcatel
N. Vaidya
Texas A&M University
October 19, 1999
Long Thin Networks
draft-montenegro-pilc-ltn-03.txt
Status of This Memo
This document is an Internet-Draft and is in full conformance
with all provisions of Section 10 of RFC2026.
This document is an individual submission to the Internet
Engineering Task Force (IETF). Comments should be submitted to the
authors or to the PILC mailing list at pilc@grc.nasa.gov.
Distribution of this memo is unlimited.
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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
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proposals along with future research items. Based on this
overview, we also recommend mechanisms for implementation in long
thin networks.
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 work 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 .................................................... 5
1.1 Network Architecture ....................................... 7
1.2 Assumptions about the Radio Link ........................... 8
2 Should it be IP or Not? ........................................ 9
2.1 Underlying Network Error Characteristics ................... 10
2.2 Non-IP Alternatives ........................................ 11
2.2.1 WAP ................................................... 11
2.2.2 Deploying Non-IP Alternatives ......................... 11
2.3 IP-based Considerations .................................... 12
2.3.1 Choosing the MTU [Stevens94, RFC1144] ................. 12
2.3.2 Path MTU Discovery [RFC1191] .......................... 12
2.3.3 Non-TCP Proposals ..................................... 13
3 The Case for TCP ................................................ 13
4 Candidate Optimizations ......................................... 14
4.1 TCP: Current Mechanisms .................................... 15
4.1.1 Slow Start and Congestion Avoidance ................... 15
4.1.2 Fast Retransmit and Fast Recovery ..................... 15
4.2 Connection Setup with T/TCP [RFC1397, RFC1644] ............. 16
4.3 Slow Start Proposals ....................................... 17
4.3.1 Larger Initial Window ................................. 17
4.3.2 Growing the Window during Slow Start .................. 18
4.3.2.1 ACK Counting ..................................... 18
4.3.2.2 ACK-every-segment ................................ 19
4.3.3 Terminating Slow Start ................................ 20
4.3.4 Generating ACKs during Slow Start ..................... 20
4.4 ACK Spacing ................................................ 20
4.5 Delayed Duplicate Acknowlegements .......................... 21
4.6 Selective Acknowledgements [RFC2018] ....................... 22
4.7 Detecting Corruption Loss .................................. 22
4.7.1 Without Explicit Notification ......................... 22
4.7.2 With Explicit Notifications ........................... 23
4.8 Active Queue Management .................................... 24
4.9 Scheduling Algorithms ...................................... 25
4.10 Split TCP and Performance-Enhancing Proxies (PEPs) ........ 26
4.10.1 Split TCP Approaches ................................. 27
4.10.2 Application Level Proxies ............................ 30
4.10.3 Snoop and its Derivatives ............................ 31
4.10.4 PEPs to handle Periods of Disconnection .............. 33
4.11 Header Compression Alternatives ........................... 34
4.12 Payload Compression ....................................... 35
4.13 TCP Control Block Interdependence [Touch97] ............... 36
5 Summary of Recommended Optimizations ............................ 37
6 Conclusion ...................................................... 39
7 Acknowledgements ................................................ 39
8 Security Considerations ......................................... 39
9 References ...................................................... 41
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Authors' addresses ................................................ 48
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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
even further. 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 Network 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 intermediate node 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
- an intermediate node (sometimes referred to as a base
station) 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 an intermediate node 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
intermediate node 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 intermediate node, 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.
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That is, full mobility implies that the devices have the
flexibility and agility to use whichever happens to be the best
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].
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.
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- What kind of reliability does the link provide? Wireless
services typically retransmit a packet (frame) until it has
been acknowledged by the target. They may allow the user to
turn off this behavior. For example, GSM allows RLP [RLP]
(Radio Link Protocol) to be turned off. Metricom has
a similar "lightweight" mode. In GSM RLP, a frame is
retransmitted until the maximum number of retransmissions
(protocol parameter) is reached. What happens when this
limit is reached is determined by the telecom operator:
the physical link connection is either disconnected or
a link reset is enforced where the sequence numbers are
resynchronized and the transmit and receive buffers are
flushed resulting in lost data. Some wireless services,
like CDMA IS95-RLP [CDMA, Karn93], limit the latency
on the wireless link by retransmitting a frame only a
couple of times. This decreases the residual frame error
rate significantly, but does not provide fully reliable
link service.
- 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 in this document that mobile devices do not transmit
and receive simultaneously.
- 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 intermediate node) 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
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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 intermediate node (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
[Stevens94, Paxson97] (and, 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.
Hence for paths that include wireless links, exclusively relying
on end-to-end mechanisms to detect and correct transmission
errors is undesireable. These should be complemented by local
link-level mechanisms. Otherwise, damaged IP packets are
propagated through the network only to be discarded at the
destination host. For example, intermediate routers are required
to check the IP header checksum, but not the UDP or TCP
checksums. Accordingly, when the payload of an IP packet is
corrupted, this is not detected until the packet arrives at its
ultimate destination.
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
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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-RTP, 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
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
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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 Considerations
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 Choosing MTU [Stevens94, RFC1144]
In slow networks, the time required to transmit the largest
possible packet may be considerable. Interactive response time
should not exceed the well-known human factors limit of 100 to
200 ms. This should be considered the maximum time budget to (1)
send a packet and (2) obtain a response. In most networking
stack implementations, (1) is highly dependent on the maximum
transmission unit (MTU). In the worst case, a small packet from
an interactive application may have to wait for a large packet
from a bulk transfer application before being sent. Hence, a
good rule of thumb is to choose an MTU such that its
transmission time is less than (or not much larger than) 200
ms.
Of course, compression and type-of-service queuing (whereby
interactive data packets are given a higher priority) may
alleviate this problem. In particular, the latter may reduce the
average wait time to about half the MTU's transmission time.
2.3.2 Path MTU Discovery [RFC1191]
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. Without Path MTU
discovery, the default IPv4 MTU size is 576. The benefits of
using a larger MTU are:
- Smaller ratio of header overhead to data
- Allows TCP to grow its congestion window faster, since
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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.
Recommendation: Path MTU discovery is recommended. [AGS98]
already recommends its use in satellite environments.
2.3.3 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:
- It is generally recognized that TCP does not perform well
in the presence of significant levels of non-congestion
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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 three-way
handshake).
- 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 the performance of TCP over long-thin
networks can be improved significantly. Mechanisms to do so are
discussed 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. Similarly, delayed ACKs
exacerbate the perceived latency on the link. Given that TCP
grows its congestion window in units of segments, small MTUs
may slow adaptation even further.
4.1.1 Slow Start and Congestion Avoidance
Slow Start and Congestion Avoidance [RFC2581] are essential
the Internet's stability. However there are two reasons why
the wireless medium adversely affects them:
- Whenever TCP's retransmission timer expires, the sender
assumes that the network is congested and invokes slow
start. This is why it is important to minimize the
losses caused by corruption, leaving only those caused
by congestion (as expected by TCP).
- The sender increases its window based on the number of ACKs
received. Their rate of arrival, of course, is dependent
on the RTT (round-trip-time) between sender and receiver,
which implies long ramp-up times in high latency links
like LTNs. The dependency lasts until the pipe is filled.
- 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 requires
link layers with low loss.
4.1.2 Fast Retransmit and Fast Recovery
When a TCP sender receives several duplicate ACKs, fast
retransmit [RFC2581] allows it to infer that a segment was lost.
The sender retransmits what it considers to be this lost
segment without waiting for the full timeout, thus saving time.
After a fast retransmit, a sender invokes the fast recovery
[RFC2581] algorithm. Fast recovery allows the sender to transmit
at half its previous rate (regulating the growth of its window
based on congestion avoidance), rather than having to begin
a slow start. This also saves time.
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In general, TCP can increase its window beyond the
delay-bandwidth product. However, in LTN links the congestion
window may remain rather small, less than four segments,
for long periods of time due to any of the following reasons:
1. Typical "file size" to be transferred over a connection
is relatively small (Web requests, Web document objects,
email messages, files, etc.) In particular, users of
LTNs are not very willing to carry out large transfers
as the response time is so long.
2. If the link has high BER, the congestion window tends
to stay small
3. When an LTN is combined with a highly congested wireline
Internet path, congestion losses on the Internet have
the same effect as 2.
4. Commonly, ISPs/operators configure only a small number
of buffers (even as few as for 3 packets) per user in
their dial-up routers
5. Often small socket buffers are recommended with LTNs in
order to prevent the RTO from inflating and to diminish
the amount of packets with competing traffic.
A small window effectively prevents the sender from taking
advantage of Fast Retransmits. Moreover, efficient recovery from
multiple losses within a single window requires adoption of new
proposals (NewReno [RFC2582]). In addition, on slow paths with
no packet reordering waiting for three duplicate ACKs to arrive
postpones retransmission unnecessarily.
Recommendation: Implement Fast Retransmit and Fast Recovery at
this time. This is a widely-implemented optimization and is
currently at Proposed Standard level. [AGS98] recommends
implementation of Fast Retransmit/Fast Recovery in satellite
environments. NewReno [RFC2582] apparently does help a sender
better handle partial ACKs and multiple losses in a single
window, but at this point is not recommended due to its
experimental nature. Instead, SACK [RFC2018] is the preferred
mechanism.
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
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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.
- 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 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.
4.3.1 Larger Initial Window
Traditional slow start, with an initial window of one segment,
is a time-consuming bandwidth adaptation procedure over
LTNs. Studies on an initial window larger than one segment
[RFC2414, AHO98] resulted in the TCP standard supporting
a maximum value of 2 [RFC2581]. Higher values are still
experimental in nature.
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
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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 2 (following RFC2581), which at least allows clients
running query-response applications to get an initial ACK from
unmodified servers without waiting for a typical delayed ACK
timeout of 200 milliseconds, and saves two round-trips. An
initial window of 3 [RFC2415] looks promising and may be
adopted in the future pending further research and experience.
4.3.2 Growing the Window 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 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
as much 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.
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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 intermediate
node.
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. For instance, a low
bandwidth link carrying acknowledgements back to the
sender, hinders the growth of the congestion window, even
if the link toward the client has a greater bandwidth
[BPK99].
Even though simulations confirm its promise (it allows
receivers to receive the second segment from unmodified
senders without waiting for a typical delayed ACK timeout
of 200 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 [RFC2581]: delayed acknowledgements
SHOULD be used by a TCP receiver.
"Disabling Delayed ACKs During Slow Start" is technically
unimplementable, as the receiver has no way of knowing 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.
However, this scheme might double ACK traffic if most
connections remain in slow-start.
Recommendation: ACK only the first segment on a new connection
with no delay.
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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. This results in the closing of the congestion window to
1 segment (which precludes fast retransmit), 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.
4.3.4 Generating ACKs during Slow Start
Mitigations that inject additional ACKs (whether
"ACK-first-segment" or "ACK-every-segment-during-slow-start")
beyond what today's conformant TCPs inject are only applicable
during the slow-start phases of a connection. After an initial
exchange, the connection usually completes slow-start, so TCPs
only inject additional ACKs when (1) the connection is closed,
and a new connection is opened, or (2) the TCPs handle idle
connection restart correctly by performing slow start.
Item (1) is typical when using HTTP/1.0, in which each
request-response transaction requires a new connection.
Persistent connections in HTTP/1.1 help in maintaining a
connection in congestion avoidance instead of constantly
reverting to slow-start. Because of this, these optimizations
which are only enabled during slow-start do not get as much
of a chance to act. Item (2), of course, is independent of
HTTP version.
4.4 ACK Spacing
During slow start, the sender responds to the incoming ACK
stream by transmitting N+1 segments for each ACK, where N is the
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number of new segments acknowledged by the incoming ACK. 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].
Furthermore, ack spacing reduces the size of the bursts.
Recommendation: No recommendation at this time. Continue
monitoring research in this area.
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. With this in mind, the Delayed Dupacks [MV97,
Vaidya99] scheme selectively delays duplicate acknowledgements
at the receiver. 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
behavior.
The Delayed Dupacks scheme can be used despite IP encryption
since the intermediate node does not need to examine the TCP
headers.
Currently, it is not well understood how long the receiver
should delay the duplicate acknowledgments. In particular, the
impact of wireless medium access control (MAC) protocol on the
choice of delay parameter needs to be studied. The MAC
protocol may affect the ability to choose the appropriate
delay (either statically or dynamically). In general,
significant variabilities in link-level retransmission times
can have an adverse impact on the performance of the Delayed
Dupacks scheme. Furthermore, as discussed later in section
4.10.3, Delayed Dupacks and some other schemes (such as Snoop
[SNOOP]) are only beneficial in certain types of network
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links.
Recommendation: Delaying duplicate acknowledgements may be
useful in specific network topologies, but a general
recommendation requires further research and experience.
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, TCP may enter congestion
avoidance as the time needed to retransmit the lost segments
may be greater than the retranmission timer.
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
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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).
Recommendation: No recommendation at this time. Schemes like
ELNR and EDDAN [MV97], in which the only systems that need to
be modified are the intermediate node and the mobile device, are
slated for adoption pending further research. However, this
solution has some limitations. Since the intermediate node must
have access to the TCP headers, the IP payload must not be
encrypted.
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.
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Recommendation: Implement ECN. In spite of this, mechanisms
for explicit corruption notification are still relevant and
should be tracked.
Note: ECN provides useful information to avoid deteriorating
further a bad situation, but has some limitations for wireless
applications. 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. In other words, the objective is to
drop m packets early to prevent n drops later on, where
m is less than n.
- 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 non-acceptable.
- Avoid lock-outs. Lack of resources in a router (and the
resultant packet drops) may, in effect, obliterate
throughput on certain connections. Because of active queue
management, it is more probable for an incoming packet to
find available buffer space at the router.
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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.
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 intermediate node. 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 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
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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 intermediate node.
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 Retransmit to resend after
the first duplicate ack and Fast Recovery to not shrink the
congestion window 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 various similar types of
performance enhancements can be employed without endangering
operations over the global Internet.
In some proposals, in addition to a PEP mechanism at the
intermediate node, 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).
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4.10.1 Split TCP Approaches
Split-TCP proposals include schemes like I-TCP [ITCP] and MTCP
[YB94] which achieve performance improvements by abandoning
end-to-end semantics.
The Mowgli architecture [MOWGLI] proposes a 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]. In addition,
the protocol provides various features that are useful with
LTNs. For example, it provides priority-based multiplexing of
concurrent connections together with shared flow control, thus
offering link capacity to interactive applications in a timely
manner even if there are bandwidth-intensive background
transfers. Also with this option, Mowgli preserves the socket
semantics on the mobile device so that legacy applications can
be run unmodified.
Employing split TCP approaches have several benefits as well as
drawbacks. Benefits related to split TCP approaches include the
following:
- Splitting the end-to-end TCP connection into two parts is a
straightforward way to shield the problems of the wireless
link from the wireline Internet path, and vice versa. Thus,
a split TCP approach enables applying local solutions to
the local problems on the wireless link. For example, it
automatically solves the problem of distinguishing congestion
related packet losses on the wireline Internet and packet
losses due to transmission error on the wireless link as these
occur on separate TCP connections. Even if both segments
experience congestion, it may be of a different nature and
may be treated as such. Moreover, temporary disconnections
of the wireless link can be effectively shielded from the
wireline Internet.
- When one of the TCP connections crosses only a single hop
wireless link or a very limited number of hops, some or all
link characteristics for the wireless TCP path are known. For
example, with a particular link we may know that the link
provides reliable delivery of packets, packets are not
delivered out of order, or the link is not subject to
congestion. Having this information for the TCP path one could
expect that defining the TCP mitigations to be employed
becomes a significantly easier task. In addition, several
mitigations that cannot be employed safely over the global
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Internet, can be successfully employed over the wireless link.
- Splitting one TCP connection into two separate ones allows
much earlier deployment of various recent proposals to improve
TCP performance over wireless links; only the TCP
implementations of the mobile device and intermediate node
need to be modified, thus allowing the vast number of Internet
hosts to continue running the legacy TCP implementations
unmodified. Any mitigations that would require modification of
TCP in these wireline hosts may take far too long to become
widely deployed.
- Allows exploitation of various application level enhancements
which may give significant performance gains (see section
4.10.2).
Drawbacks related to split TCP approaches include the
following:
- One of the main criticisms against the split TCP approaches is
that it breaks TCP end-to-end semantics. This has various
drawbacks some of which are more severe than others. The most
detrimental drawback is probably that splitting the TCP
connection disables end-to-end usage of IP layer security
mechanisms, precluding the application of IPSec to achieve
end-to-end security. Still, IPSec could be employed separately
in each of the two parts, thus requiring the intermediate node
to become a party to the security association between the
mobile device and the remote host. This, however, is an
undesireable or unacceptable alternative in most cases. Other
security mechanisms above the transport layer, like TLS
[RFC2246] or SOCKS [RFC1928], should be employed for
end-to-end security.
- Another drawback of breaking end-to-end semantics is that
crashes of the intermediate node become unrecoverable
resulting in termination of the TCP connections. Whether this
should be considered a severe problem depends on the expected
frequency of such crashes.
- In many occasions claims have been stated that if TCP
end-to-end semantics is broken, applications relying on TCP to
provide reliable data delivery become more vulnerable. This,
however, is an overstatement as a well-designed application
should never fully rely on TCP in achieving end-to-end
reliability at the application level. First, current APIs to
TCP, such as the Berkeley socket interface, do not allow
applications to know when an TCP acknowledgement for
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previously sent user data arrives at TCP sender. Second, even
if the application is informed of the TCP acknowledgements,
the sending application cannot know whether the receiving
application has received the data: it only knows that the data
reached the TCP receive buffer at the receiving end. Finally,
in order to achieve end-to-end reliability at the application
level an application level acknowledgement is required to
confirm that the receiver has taken the appropriate actions on
the data it received.
- When a mobile device moves, it is subject to handovers by the
serving base station. If the base station acts as the
intermediate node for the split TCP connection, the state of
both TCP endpoints on the previous intermediate node must be
transferred to the new intermediate node to ensure continued
operation over the split TCP connection. This requires extra
work and causes overhead. However, in most of the W-WAN
wireless networks, unlike in W-LANs, the W-WAN base station
does not provide the mobile device with the connection point
to the wireline Internet (such base stations may not even have
an IP stack). Instead, the W-WAN network takes care of the
mobility and retains the connection point to the wireline
Internet unchanged while the mobile device moves. Thus, TCP
state handover is not required in most W-WANs.
- The packets traversing through all the protocol layers up to
transport layer and again down to the link layer result in
extra overhead at the intermediate node. In case of LTNs with
low bandwidth, this extra overhead does not cause serious
additional performance problems unlike with W-LANs that
typically have much higher bandwidth.
- Split TCP proposals are not applicable to networks with
asymmetric routing. Deploying a split TCP approach requires
that traffic to and from the mobile device be routed through
the intermediate node. With some networks, this cannot be
accomplished, or it requires that the intermediate node is
located several hops away from the wireless network edge which
in turn is unpractical in many cases and may result in
non-optimal routing.
- Split TCP, as the name implies, does not address problems
related to UDP.
It should noted that using split TCP does not necessarily
exclude simultaneous usage of IP for end-to-end connectivity.
Correct usage of split TCP should be managed per application or
per connection and should be under the end-user control so that
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the user can decide whether a particular TCP connection or
application makes use of split TCP or whether it operates
end-to-end directly over IP.
Recommendation: Split TCP proposals that alter TCP semantics are
not recommended. Deploying custom protocols on the wireless
link, such as MOWGLI proposes is not recommended, because this
note gives preference to (1) improving TCP instead of designing
a custom protocol and (2) allowing end-to-end sessions at all
times.
4.10.2 Application Level Proxies
Nowadays, application level proxies are widely used in the
Internet. Such proxies include Web proxy caches, relay MTAs
(Mail Transfer Agents), and secure transport proxies (e.g.,
SOCKS). In effect, employing an application level proxy results
in a "split TCP connection" with the proxy as the intermediary.
Hence, some of the problems present with wireless links, such as
combining of a congested wide-area Internet path with a wireless
LTN link, are automatically alleviated to some extent.
The application protocols often employ plenty of (unnecessary)
round trips, lots of headers and inefficient encoding. Even
unnecessary data may get delivered over the wireless link in
regular application protocol operation. In many cases a
significant amount of this overhead can be reduced by simply
running an application level proxy on the intermediate node.
With LTN links, significant additional improvement can be
achieved by introducing application level proxies with
application-specific enhancements. Such a proxy may employ an
enhanced version of the application protocol over the wireless
link. In an LTN environment enhancements at the application
layer may provide much more notable performance improvements
than any transport level enhancements.
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 intermediate
node. Such a pair may be either explicit or fully transparent to
the applications, but it is, at all times, under the end-user
control. Good examples of enhancements achieved with
application-specific proxies include Mowgli WWW [LAKLR95],
[LHKR96] and WebExpress [HL96], [CTCSM97].
Recommendation: Usage of application level proxies is
conditionally recommended: an application must be proxy enabled
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and the decision of employing a proxy for an application must be
under the user control at all times.
4.10.3 Snoop and its Derivatives
Berkeley's SNOOP protocol [SNOOP] is a hybrid scheme mixing
link-layer reliability mechanisms with the split connection
approach. It is an improvement over split TCP approaches in that
end-to-end semantics are retained. 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.
Thus, the Snoop protocol is designed to avoid unnecessary fast
retransmits by the TCP sender, when the wireless link layer
retransmits a packet locally. Consider a system that does not
use the Snoop agent. Consider a TCP sender S that sends packets
to receiver R via an intermediate node IN. Assume that the
sender sends packet A, B, C, D, E (in that order) which are
forwarded by IN to the wireless receiver R. Assume that the
intermediate node then retransmits B subsequently, because the
first transmission of packet B is lost due to errors on the
wireless link. In this case, receiver R receives packets A, C,
D, E and B (in that order). Receipt of packets C, D and E
triggers duplicate acknowledgements. When the TCP sender
receives three duplicate acknowledgements, it triggers fast
retransmit (which results in a retransmission, as well as
reduction of congestion window). The fast retransmit occurs
despite the link level retransmit on the wireless link,
degrading throughput.
SNOOP [SNOOP] deals with this problem by dropping TCP dupacks
appropriately (at the intermediate node). The Delayed Dupacks
(see section 4.5) attempts to approximate Snoop without
requiring modifications at the intermediate node. Such schemes
are needed only if the possibility of a fast retransmit due to
wireless errors is non-negligible. In particular, if the
wireless link uses a stop-and-go protocol (or otherwise delivers
packets in-order), then these schemes are not very beneficial.
Also, if the bandwidth-delay product of the wireless link is
smaller than four segments, the probability that the
intermediate node will have an opportunity to send three new
packets before a lost packet is retransmitted is small. Since
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at least three dupacks are needed to trigger a fast retransmit,
with a wireless bandwidth-delay product less than four packets,
schemes such as Snoop and Delayed Dupacks would not be necessary
(unless the link layer is not designed properly). Conversely,
when the wireless bandwidth-delay product is large enough, Snoop
can provide significant performance improvement (compared with
standard TCP). For further discussion on these topics, please
refer to [Vaidya99].
The Delayed Dupacks scheme tends to provide performance benefit
in environments where Snoop performs well. In general,
performance improvement achieved by the Delayed Dupacks scheme
is a function of packet loss rates due to congestion and
transmission errors. When congestion-related losses occur, the
Delayed Dupacks scheme unnecessarily delays retransmission.
Thus, in the presence of congestion losses, the Delayed Dupacks
scheme cannot achieve the same performance improvement as Snoop.
However, simulation results [Vaidya99] indicate that the Delayed
Dupacks can achieve a significant improvement in performance
despite moderate congestion losses.
WTCP [WTCP] is similar to SNOOP in that it preserves end-to-end
semantics. In WTCP, the intermediate node uses a complex scheme
to hide the time it spends recovering from local errors across
the wireless link (this typically includes retransmissions due
to error recovery, but may also include time spent dealing with
congestion). The idea is for the sender to derive a smooth
estimate of round-trip time. 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 intermediate node.
SNOOP and WTCP require the intermediate node to examine and
operate on the traffic between the portable wireless device and
the TCP server on the wired Internet. SNOOP and WTCP do not work
if the IP traffic is encrypted, unless, of course, the
intermediate node 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
intermediate node. Furthermore, if the intermediate node
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.
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Encryption of IP packets via IPSEC's ESP header (in either
transport or tunnel mode) renders the TCP header and payload
unintelligible to the intermediate node. This precludes SNOOP
(and WTCP) from working, because it needs to examine the TCP
headers in both directions. Possible solutions involve:
- making the SNOOP (or WTCP) intermediate node a party to the
security association between the client and the server
- IPSEC tunneling mode, terminated at the SNOOPing intermediate
node
However, these techniques require that users trust intermediate
nodes. Users valuing both privacy and performance should use
SSL or SOCKS for end-to-end security. These, however, are
implemented above the transport layer, and are not as resistant
to some security attacks (for example, those based on guessing
TCP sequence numbers) as IPSEC.
Recommendation: Implement SNOOP on intermediate nodes now.
Research results are encouraging, and it is an "invisible"
optimization in that neither the client nor the server needs to
change, only the intermediate node (for basic SNOOP without
SACK). However, as discussed above there is little or no benefit
from implementing SNOOP if:
1. The wireless link provides reliable, in-order packet
delivery, or,
2. The bandwidth-delay product of the wireless link is
smaller than four segments.
4.10.4 PEPs to handle Periods of Disconnection
Periods of disconnection are very common in wireless networks,
either during handoff, due to lack of resources (dropped
connections) or natural obstacles. During these periods, a TCP
sender does not receive the expected acknowledgements. Upon
expiration of the retransmit timer, this causes TCP to close its
congestion window with all the related drawbacks.
Re-transmitting packets is useless since the connection is
broken. [M-TCP] aims at enabling TCP to better handle handoffs
and periods of disconnection, while preserving end-to-end
semantics. M-TCP adds an element: supervisor host (SH-TCP) at
the edge of the wireless network.
This intermediate node monitors the traffic coming from the
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sender to the mobile device. It does not break end-to-end
semantics because the ACKs sent from the intermediate node to
the sender are effectively the ones sent by the mobile node. The
principle is to generally leave the last byte unacknowledged.
Hence, SH-TCP could shut down the sender's window by sending
the ACK for the last byte with a window set to zero. Thus the
sender will go to persist mode.
The second optimization is done on both the intermediate node
and the mobile host. On the latter, TCP is aware of the current
state of the connection. In the event of a disconnection, it is
capable of freezing all timers. Upon reconnection, the mobile
sends a specially marked ACK with the number of the highest byte
received. The intermediate node assumes that the mobile is
disconnected because it monitors the flow on the wireless link,
so in the absence of acknowledgments from the mobile, it will
inform SH-TCP, which will send the ACK closing the sender window
as described in the previous paragraph. The intermediate node
learns that the mobile is again connected when it receives a
duplicate acknowledgment marked as reconnected. At this point
it sends a duplicate ACK to the sender and grows the window.
The sender exits persist mode and resumes transmitting at
the same rate as before. It begins by retransmitting any data
previously unacknowledged by the mobile node. Non overlapping
or non soft handoffs are lightweight because the previous
intermediate system can shrink the window, and the new one
modifies it as soon as it has received an indication from
the mobile.
Recommendation: M-TCP is not slated for adoption at this moment,
because of the higly experimental nature of the proposal, and
the uncertainty that TCP/IP implementations handle zero window
updates correctly. Continue tracking developments in this space.
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-RTP, IPHC-PPP] provide the following
benefits:
- Improve interactive response time
- Allow using small packets for bulk data with good line
efficiency
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- Allow using small packets for delay sensitive low
data-rate traffic
- Decrease header overhead (for a common TCP segment size of
512 the header overhead of IPv4/TCP within a Mobile IP
tunnel can decrease from 11.7 to less than 1 per cent.
- Reduce packet loss rate over lossy links (because of the
smaller cross-section of compressed packets).
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 transmission of uncompressed
headers to allow resynchronization without waiting for a TCP
timeout (and executing congestion avoidance procedures).
Recommendation: Implement [IPHC], in particular as it relates
to IP-in-IP [RFC2003] and Minimal Encapsulation [RFC2004] for
Mobile IP, as well as TCP header compression for lossy links
and links that reorder packets. PPP capable devices should
implement [IPHC-PPP]. VJ header compression may optionally
be implemented as it is a widely deployed Proposed Standard.
However, it should only be enabled when operating over
reliable LTNs, because even a single bit error most probably
would result in a full TCP window being dropped, followed by
a costly recovery via slow-start.
4.12 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/1.1 already supports compression of the message body.
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For example, to use zlib compression the relevant directives
are: "Content-Encoding: deflate" and "Accept-Encoding:
deflate" [HTTP-PERF].
HTTP-NG is considering supporting compression of resources at the
HTTP level, which would provide equivalent benefits for common
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: IPComp may optionally be implemented. Track
HTTP-NG standardization and deployment for now. HTTP/1.1
compression using zlib SHOULD be implemented.
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 sharing
information amongst systems in a LAN not just within a given
system. [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: This mechanism is recommended, although HTTP/1.1
with its persistent connections may partially achieve the same
effect without it. Other applications (even HTTP/1.0) may find
it useful. Continue monitoring research on this. In particular,
work on a "Congestion Manager" [CM] may generalize this concept
of sharing information among protocols and applications with
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a view to making them more adaptable to network conditions.
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
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 unknown chances
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 intermediate node. 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), IN
(intermediate node) and NI (network infrastructure). These
entities are to be understood within the context of Section 1.1
("Network Architecture"). NA simply means "not applicable."
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 2581 (PS) WS Yes
Window (initial_window=2)
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 IN (see 4.11)
IP Payload RFC 2393 (PS) WD Yes
Compression (simultaneously
(IPComp) needed on Server)
Header RFC 2507 (PS), WD Yes
Compression RFC 2509 (PS) IN (For IPv4, TCP and
Mobile IP, PPP)
SNOOP plus SACK In limited use IN Yes
WD (for SACK)
Fast retransmit/fast RFC 2581 (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 IN (for
notifications)
Class-based Queuing NA WD When stable
on End Systems
Explicit Congestion RFC 2481 (EXP) WD Yes
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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. The following individuals have also provided
valuable feedback: Mark Allman (NASA), Vern Paxson (ACIRI),
Raphi Rom (Technion/Sun), Charlie Perkins (Nokia), Peter Stark
(Phone.com).
8 Security Considerations
The mechanisms discussed and recommended in this document have
been proposed in previous publications. The security
considerations outlined in the original discussions apply here
as well. Several security issues are also discussed throughout
this document. Additionally, we present below a non-exhaustive
list of the most salient issues concerning our recommended
mechanisms:
- Larger Initial TCP Window Size
No known security issues [RFC2414, RFC2581].
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- Header Compression
May be open to some denial of service attacks. But any
attacker in a position to launch these attacks would have
much stronger attacks at his disposal [IPHC, IPHC-RTP].
- Congestion Control, Fast Retransmit/Fast Recovery
An attacker may force TCP connections to grind to a halt,
or, more dangerously, behave more aggressively. The latter
possibility may lead to congestion collapse, at least in
some regions of the network [RFC2581].
- Explicit Congestion Notification
It does not appear to increase the vulnerabilities in the
network. On the contrary, it may reduce them by aiding in
the identification of flows unresponsive to or non-compliant
with TCP congestion control [ECN].
- Sharing of Network Performance Information (TCP Control Block
Sharing and Congestion Manager module)
Some information should not be shared. For example, TCP
sequence numbers are used to protect against spoofing attacks.
Even limiting the sharing to performance values leaves open
the possibility of denial-of-service attacks [Touch97].
- Performance Enhancing Proxies
These systems are men-in-the-middle from the point of view of
their security vulnerabilities. Accordingly, they must be used
with extreme care so as to prevent their being hijacked and
misused.
This last point is not to be underestimated: there is a general
security concern whenever an intermediate node performs
operations different from those carried out in an end-to-end
basis. This is not specific to performance-enhancing proxies.
In particular, there may be a tendency to forgo IPSEC-based
privacy in order to allow, for example, a SNOOP module, header
compression (TCP, UDP, RTP, etc), or HTTP proxies to work.
Adding end-to-end security at higher layers (for example via RTP
encryption, or via TLS encryption of the TCP payload) alleviates
the problem. However, this still leaves protocol headers in the
clear, and these may be exploited for traffic analysis and
denial-of-service attacks.
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Prud'hommeaux (W3C, MIT), H…kon Lie (W3C, INRIA), Chris Lilley
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[IPHC] Mikael Degermark, Bjorn Nordgren, Stephen Pink. "IP
Header Compression," RFC 2507, February 1999.
[IPHC-RTP] S. Casner, V. Jacobson. "Compressing IP/UDP/RTP
Headers for Low-Speed Serial Links," RFC 2508, February 1999.
[IPHC-PPP] Mathias Engan, S. Casner, C. Bormann. "IP Header
Compression over PPP," RFC 2509, February 1999.
[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,
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[KRLKA97] Kojo, M., Raatikainen, K., Liljeberg, M., Kiiskinen,
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Telephone Links," in IEEE Journal on Selected Areas of
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Raatikainen, K., "Optimizing World-Wide Web for Weakly-Connected
Mobile Workstations: An Indirect Approach," in Proc. 2nd Int.
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"Mowgli WWW Software: Improved Usability of WWW in Mobile WAN
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Control for Improving Wireless Link Throughput, Range, and Energy
Efficiency," Proc. IEEE INFOCOM'98, April 1998.
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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
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Macroscopic Behavior of the TCP Congestion Avoidance Algorithm,"
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[MTCP] Brown, K. Singh, S., "A Network Architecture for Mobile
Computing," Proc. IEEE INFOCOM'96, pp. 1388-1396, March 1996.
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[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.
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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.
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[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.
[RLP] ETSI, "Radio Link Protocol for Data and Telematic Services
on the Mobile Station - Base Station System (MS-BSS) interface
and the Base Station System - Mobile Switching Center (BSS-MSC)
interface," GSM Specification 04.22, Version 3.7.0, February
1992.
[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)",
RFC 1661, July 1994.
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[RFC1928] M. Leech, M. Ganis, Y. Lee, R. Kuris, D. Koblas,
L. Jones, "SOCKS Protocol Version 5", RFC 1928, March 1996.
[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.
[RFC2002] Perkins, C., Editor, "IP Mobility Support," RFC 2002,
October 1996.
[RFC2003] Perkins, C., Editor, "IP Encapsulation within IP,"
RFC 2003, October 1996.
[RFC2004] Perkins, C., Editor, "Minimal Encapsulation within
IP," RFC 2004, 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.
[RFC2246] T. Dierk, E. Allen, "TLS Protocol Version 1", RFC
2246, January 1999.
[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.
[RFC2581] M. Allman, V. Paxson, W. Stevens, "TCP Congestion
Control," April 1999. RFC 2581.
[RFC2582] Floyd, S., Henderson, T., "The NewReno Modification to
TCP's Fast Recovery Algorithm," April 1999. RFC 2582.
[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.
Expires April 19, 2000 [Page 46]
INTERNET DRAFT Long Thin Networks October 1999
[Stevens94] R. Stevens, "TCP/IP Illustrated, Volume 1,"
Addison-Wesley, 1994 (section 2.10 for MTU size considerations
and section 11.3 for weak checksums).
[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.
[Vaidya99] N. H. Vaidya, M. Mehta, C. Perkins, G. Montenegro,
"Delayed Duplicate Acknowledgements: A TCP-Unaware Approach to
Improve Performance of TCP over Wireless," Technical Report
99-003, Computer Science Dept., Texas A&M University, February
1999.
[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.
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INTERNET DRAFT Long Thin Networks October 1999
Authors' addresses
Questions about this document may be directed at:
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
Expires April 19, 2000 [Page 48]
INTERNET DRAFT Long Thin Networks October 1999
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
Nitin Vaidya
Dept. of Computer Science
Texas A&M University
College Station, TX 77843-3112
Phone: 409-845-0512
Fax: 409-847-8578
Email: vaidya@cs.tamu.edu
Expires April 19, 2000 [Page 49]
