path: root/doc/rfc/rfc2757.txt

Network Working Group                                      G. Montenegro
Request for Comments: 2757                        Sun Microsystems, Inc.
Category: Informational                                       S. Dawkins
                                                         Nortel Networks
                                                                 M. Kojo
                                                  University of Helsinki
                                                               V. Magret
                                                                 Alcatel
                                                               N. Vaidya
                                                    Texas A&M University
                                                            January 2000


                           Long Thin Networks

Status of this Memo

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

Copyright Notice

   Copyright (C) The Internet Society (2000).  All Rights Reserved.

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.

   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 .................................................    3
      1.1 Network Architecture ....................................    5
      1.2 Assumptions about the Radio Link ........................    6
   2 Should it be IP or Not?  .....................................    7
      2.1 Underlying Network Error Characteristics ................    7
      2.2 Non-IP Alternatives .....................................    8
         2.2.1 WAP ................................................    8
         2.2.2 Deploying Non-IP Alternatives ......................    9
      2.3 IP-based Considerations .................................    9
         2.3.1 Choosing the MTU [Stevens94, RFC1144] ..............    9
         2.3.2 Path MTU Discovery [RFC1191] .......................   10
         2.3.3 Non-TCP Proposals ..................................   10
   3 The Case for TCP .............................................   11
   4 Candidate Optimizations ......................................   12
      4.1 TCP: Current Mechanisms .................................   12
         4.1.1 Slow Start and Congestion Avoidance ................   12
         4.1.2 Fast Retransmit and Fast Recovery ..................   12
      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 Growing the Window 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
         4.3.4 Generating ACKs during 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 ........................   20
      4.8 Active Queue Management .................................   21
      4.9 Scheduling Algorithms ...................................   21
      4.10 Split TCP and Performance-Enhancing Proxies (PEPs) .....   22
         4.10.1 Split TCP Approaches ..............................   23
         4.10.2 Application Level Proxies .........................   26
         4.10.3 Snoop and its Derivatives .........................   27
         4.10.4 PEPs to handle Periods of Disconnection ...........   29
      4.11 Header Compression Alternatives ........................   30
      4.12 Payload Compression ....................................   31
      4.13 TCP Control Block Interdependence [Touch97] ............   32
   5 Summary of Recommended Optimizations .........................   33
   6 Conclusion ...................................................   35
   7 Acknowledgements .............................................   35
   8 Security Considerations ......................................   35




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   9 References ...................................................   36
   Authors' Addresses .............................................   44
   Full Copyright Statement .......................................   46

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 desirable to forego 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
   desirability 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.  That
   is, full mobility implies that the devices have the flexibility and
   agility to use whichever happens to be the best network connection



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   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.

      -  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.



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      -  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 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



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   wireless links, exclusively relying on end-to-end mechanisms to
   detect and correct transmission errors is undesirable. 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
   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





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   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 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 the 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.







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   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 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]




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      -  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 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 preferable 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.



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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].

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.




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   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.

   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.




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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 successfully.  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.







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   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.

   [RFC2581] now allows an initial congestion window of two segments. A
   larger initial window, perhaps as many as four segments, might be
   allowed in the 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 behind 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 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.



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   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 preferable 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 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




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   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 retransmission
   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
   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.








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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.

   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.






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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.

   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:




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      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
   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.






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   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).

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.





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   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
      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



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      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 undesirable 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 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



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      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 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.



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   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 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.





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   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 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



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   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.

   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



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   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 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 highly 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.  For
   example, to use zlib compression the relevant directives are:
   "Content-Encoding: deflate" and "Accept-Encoding:  deflate" [HTTP-
   PERF].



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   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. Implementing HTTP/1.1
   compression using zlib SHOULD is recommended.

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 a view to making
   them more adaptable to network conditions.







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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.

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




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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

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.



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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].

   -  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].




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   -  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 forego 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.

9 References

   [ACKSPACING]   Partridge, C., "ACK Spacing for High Delay-Bandwidth
                  Paths with Insufficient Buffering", Work in Progress.

   [ADGGHOSSTT98] Allman, M., Dawkins, S., Glover, D., Griner, J.,
                  Henderson, T., Heidemann, J., Kruse, H., Osterman, S.,
                  Scott, K., Semke, J., Touch, J. and D. Tran, "Ongoing
                  TCP Research Related to Satellites", Work in Progress.

   [AGS98]        Allman, M., Glover, D. and L. Sanchez, "Enhancing TCP
                  Over Satellite Channels using Standard Mechanisms",
                  BCP 28, RFC 2488, January 1999.









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   [Allman98]     Mark Allman. On the Generation and Use of TCP
                  Acknowledgments. ACM Computer Communication Review,
                  28(5), October 1998.

   [AHO98]        Allman, M., Hayes, C., Ostermann, S., "An Evaluation
                  of TCP with Larger Initial Windows," Computer
                  Communication Review, 28(3), July 1998.

   [BBKT96]       Bhagwat, P., Bhattacharya, P., Krishna, A., Tripathi,
                  S., "Enhancing Throughput over Wireless LANs Using
                  Channel State Dependent Packet Scheduling," in Proc.
                  IEEE INFOCOM'96, pp. 1133-40, March 1996.

   [BBKVP96]      Bakshi, B., P., Krishna, N., Vaidya, N., Pradhan,
                  D.K., "Improving Performance of TCP over Wireless
                  Networks," Technical Report 96-014, Texas A&M
                  University, 1996.

   [BPSK96]       Balakrishnan, H., Padmanabhan, V., Seshan, S., Katz,
                  R., "A Comparison of Mechanisms for Improving TCP
                  Performance over Wireless Links," in ACM SIGCOMM,
                  Stanford, California, August 1996.

   [BPK99]        Balakrishnan, H., Padmanabhan, V., Katz, R., "The
                  effects of asymmetry on TCP performance," ACM Mobile
                  Networks and Applications (MONET), Vol. 4, No. 3,
                  1999, pp. 219-241.

   [BV97]         S. Biaz and N. H. Vaidya, "Distinguishing Congestion
                  Losses  from Wireless Transmission Losses: A Negative
                  Result," Seventh International Conference on Computer
                  Communications and Networks (IC3N), New Orleans,
                  October 1998.

   [BV98]         Biaz, S., Vaidya, N., "Sender-Based heuristics for
                  Distinguishing Congestion Losses from Wireless
                  Transmission Losses," Texas A&M University, Technical
                  Report 98-013, June 1998.

   [BV98a]        Biaz, S., Vaidya, N., "Discriminating Congestion
                  Losses from Wireless Losses using Inter-Arrival Times
                  at the Receiver," Texas A&M University, Technical
                  Report 98-014, June 1998.

   [BW97]         Brasche, G., Walke, B., "Concepts, Services, and
                  Protocols of the New GSM Phase 2+ general Packet Radio
                  Service," IEEE Communications Magazine, Vol. 35, No.
                  8, August 1997.



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   [CB96]         Cheshire, S., Baker, M., "Experiences with a Wireless
                  Network in MosquitoNet," IEEE Micro, February 1996.
                  Available online as:
                  http://rescomp.stanford.edu/~cheshire/papers
                  /wireless.ps.

   [CDMA]         Electronic Industry Alliance(EIA)/Telecommunications
                  Industry Association (TIA), IS-95: Mobile Station-Base
                  Station Compatibility Standard for Dual-Mode Wideband
                  Spread Spectrum Cellular System, 1993.

   [CDPD]         Wireless Data Forum, CDPD System Specification,
                  Release 1.1, 1995.

   [CM]           Hari Balakrishnan and Srinivasan Seshan, "The
                  Congestion Manager," Work in Progress.

   [CTCSM97]      Chang, H., Tait, C., Cohen, N., Shapiro, M.,
                  Mastrianni, S., Floyd, R., Housel, B., Lindquist, D.,
                  "Web Browsing in a Wireless Environment: Disconnected
                  and Asynchronous Operation in ARTour Web Express," in
                  Proc. MobiCom'97, Budapest, Hungary, September 1997.

   [Demers90]     Demers, A., Keshav, S., and Shenker, S., Analysis and
                  Simulation of a Fair Queueing Algorithm,
                  Internetworking: Research and Experience, Vol. 1,
                  1990, pp. 3-26.

   [ECN]          Ramakrishnan, K. and S. Floyd, "A Proposal to add
                  Explicit Congestion Notification (ECN) to IP", RFC
                  2481, January 1999.

   [Floyd95]      Floyd, S., and Jacobson, V., Link-sharing and Resource
                  Management Models for Packet Networks. IEEE/ACM
                  Transactions on Networking, Vol. 3 No. 4, pp. 365-386,
                  August 1995.

   [FSS98]        Fragouli, C., Sivaraman, V., Srivastava, M.,
                  "Controlled Multimedia Wireless Link Sharing via
                  Enhanced Class-Based Queueing with Channel-State-
                  Dependent Packet Scheduling," Proc. IEEE INFOCOM'98,
                  April 1998.

   [GPRS]         ETSI, "General Packet Radio Service (GPRS): Service
                  Description, Stage 2," GSM03.60, v.6.1.1 August 1998.






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   [GSM]          Rahnema, M., "Overview of the GSM system and protocol
                  architecture," IEEE Communications Magazine, vol. 31,
                  pp 92-100, 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.

   [HTTP-PERF]    Henrik Frystyk Nielsen (W3C, MIT), Jim Gettys (W3C,
                  Digital), Anselm Baird-Smith (W3C, INRIA), Eric
                  Prud'hommeaux (W3C, MIT), Hon Lie (W3C, INRIA), Chris
                  Lilley (W3C, INRIA), "Network Performance Effects of
                  HTTP/1.1, CSS1, and PNG," ACM SIGCOMM '97, Cannes,
                  France, September 1997.  Available at:
                  http://www.w3.org/Protocols/HTTP/Performance
                  /Pipeline.html

   [IPPCP]        Shacham, A., Monsour, R., Pereira, R. and M. Thomas,
                  "IP Payload Compression Protocol (IPComp)", RFC 2393,
                  December 1998.

   [IPHC]         Degermark, M., Nordgren, B. and S. Pink, "IP Header
                  Compression", RFC 2507, February 1999.

   [IPHC-RTP]     Casner, S. and  V. Jacobson, "Compressing IP/UDP/RTP
                  Headers for Low-Speed Serial Links", RFC 2508,
                  February 1999.

   [IPHC-PPP]     Engan, M., Casner, S. and 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, 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.

   [Karn93]       Karn, P., "The Qualcomm CDMA Digital Cellular System"
                  Proc. USENIX Mobile and Location-Independent Computing
                  Symposium, USENIX Association, August 1993.






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   [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)", 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
                  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




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   [M-TCP]        Brown, K. Singh, S., "M-TCP: TCP for Mobile Cellular
                  Networks," ACM Computer Communications Review Vol.
                  27(5), 1997.  Available at
                  ftp://ftp.ece.orst.edu/pub/singh/papers/mtcp.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]       White, J., "NETBLT (Network Block Transfer Protocol)",
                  Work in Progress.

   [Paxson97]     V. Paxson, "End-to-End Internet Packet Dynamics,"
                  Proc. SIGCOMM '97.  Available at
                  ftp://ftp.ee.lbl.gov/papers/vp-pkt-dyn-sigcomm97.ps.Z

   [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., Shenker, S., Wroclawski, J. and L.
                  Zhang, "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. and J. Sax, "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", STD 3, RFC 1122, 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.





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   [RFC1191]      Mogul, J. and S. Deering, "Path MTU Discovery", RFC
                  1191, November 1990.

   [RFC1397]      Braden, R., "Extending TCP for Transactions --
                  Concepts", RFC 1397, November 1992.

   [RFC1644]      Braden, R., "T/TCP -- TCP Extensions for Transactions
                  Functional Specification", RFC 1644, July 1994.

   [RFC1661]      Simpson, W., "The Point-To-Point Protocol (PPP)", STD
                  51, RFC 1661, July 1994.

   [RFC1928]      Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D.
                  and L. Jones, "SOCKS Protocol Version 5", RFC 1928,
                  March 1996.

   [RFC1986]      Polites, W., Wollman, W., Woo, D. and R. Langan,
                  "Experiments with a Simple File Transfer Protocol for
                  Radio Links using Enhanced Trivial File Transfer
                  Protocol (ETFTP)", RFC 1986, August 1996.

   [RFC2002]      Perkins, C., "IP Mobility Support", RFC 2002, October
                  1996.

   [RFC2003]      Perkins, C., "IP Encapsulation within IP", RFC 2003,
                  October 1996.

   [RFC2004]      Perkins, C., "Minimal Encapsulation within IP", RFC
                  2004, October 1996.

   [RFC2018]      Mathis, M., Mahdavi, J., Floyd, S. and A. Romanow,
                  "TCP Selective Acknowledgment Options", RFC 2018,
                  October 1996.

   [RFC2188]      Banan, M., Taylor, M. and J. Cheng, "AT&T/Neda's
                  Efficient Short Remote Operations (ESRO) Protocol
                  Specification Version 1.2", RFC 2188, September 1997.

   [RFC2246]      Dierk, T. and E. Allen, "TLS Protocol Version 1", RFC
                  2246, January 1999.

   [RFC2414]      Allman, M., Floyd, S. and C. Partridge. "Increasing
                  TCP's Initial Window", RFC 2414, September 1998.

   [RFC2415]      Poduri, K.and K. Nichols, "Simulation Studies of
                  Increased Initial TCP Window Size", RFC 2415,
                  September 1998.




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   [RFC2416]      Shepard, T. and C. Partridge, "When TCP Starts Up With
                  Four Packets Into Only Three Buffers", RFC 2416,
                  September 1998.

   [RFC2581]      Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
                  Control", RFC 2581, April 1999.

   [RFC2582]      Floyd, S. and T. Henderson, "The NewReno Modification
                  to TCP's Fast Recovery Algorithm", RFC 2582, April
                  1999.

   [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.

   [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]        Jacobson, V., Braden, R. and D. Borman, "TCP
                  Extensions for High Performance", RFC 1323, May 1992.

   [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/






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   [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:

   Gabriel E. Montenegro
   Sun Labs Networking and Security Group
   Sun Microsystems, Inc.
   901 San Antonio Road
   Mailstop UMPK 15-214
   Mountain View, California 94303

   Phone: +1-650-786-6288
   Fax:   +1-650-786-6445
   EMail: gab@sun.com


   Spencer Dawkins
   Nortel Networks
   P.O. Box 833805
   Richardson, Texas 75083-3805

   Phone: +1-972-684-4827
   Fax:   +1-972-685-3292
   EMail: sdawkins@nortel.com










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   Markku Kojo
   Department of Computer Science
   University of Helsinki
   P.O. Box 26 (Teollisuuskatu 23)
   FIN-00014 HELSINKI
   Finland

   Phone: +358-9-1914-4179
   Fax:   +358-9-1914-4441
   EMail: 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

   Phone: +1-972-996-2625
   Fax:   +1-972-996-5902
   EMail: vincent.magret@aud.alcatel.com


   Nitin Vaidya
   Dept. of Computer Science
   Texas A&M University
   College Station, TX 77843-3112

   Phone: 979-845-0512
   Fax: 979-847-8578
   EMail: vaidya@cs.tamu.edu


















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Full Copyright Statement

   Copyright (C) The Internet Society (2000).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
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Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.



















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