TitleCompressing TCP/IP Headers for Low-Speed Serial Links
AuthorV. Jacobson
DateFebruary 1990

   Network Working Group                                     V. Jacobson/1/
   Request for Comments: 1144                                           LBL
                                                              February 1990

                          Compressing TCP/IP Headers

                          for Low-Speed Serial Links

   Status of this Memo

   This RFC is a proposed elective protocol for the Internet community and
   requests discussion and suggestions for improvement.  It describes a
   method for compressing the headers of TCP/IP datagrams to improve
   performance over low speed serial links.  The motivation, implementation
   and performance of the method are described.  C code for a sample
   implementation is given for reference.  Distribution of this memo is

   NOTE: Both ASCII and Postscript versions of this document are available.
         The ASCII version, obviously, lacks all the figures and all the
	 information encoded in typographic variation (italics, boldface,
	 etc.).  Since this information was, in the author's opinion, an
	 essential part of the document, the ASCII version is at best
	 incomplete and at worst misleading.  Anyone who plans to work
	 with this protocol is strongly encouraged obtain the Postscript
	 version of this RFC.


   1  Introduction                                                        1

   2  The problem                                                         1

   3  The compression algorithm                                           4

      3.1 The basic idea . . . . . . . . . . . . . . . . . . . . . . . .  4

      3.2 The ugly details . . . . . . . . . . . . . . . . . . . . . . .  5

         3.2.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . .  5

         3.2.2 Compressed packet format. . . . . . . . . . . . . . . . .  7

         3.2.3 Compressor processing . . . . . . . . . . . . . . . . . .  8

         3.2.4 Decompressor processing . . . . . . . . . . . . . . . . . 12

   4  Error handling                                                     14

      4.1 Error detection  . . . . . . . . . . . . . . . . . . . . . . . 14

      4.2 Error recovery . . . . . . . . . . . . . . . . . . . . . . . . 17

   5  Configurable parameters and tuning                                 18

      5.1 Compression configuration  . . . . . . . . . . . . . . . . . . 18

      5.2 Choosing a maximum transmission unit . . . . . . . . . . . . . 20

      5.3 Interaction with data compression  . . . . . . . . . . . . . . 21

   6  Performance measurements                                           23

   7  Acknowlegements                                                    25

   A  Sample Implementation                                              27

      A.1 Definitions and State Data . . . . . . . . . . . . . . . . . . 28

      A.2 Compression  . . . . . . . . . . . . . . . . . . . . . . . . . 31

      A.3 Decompression  . . . . . . . . . . . . . . . . . . . . . . . . 37

      A.4 Initialization . . . . . . . . . . . . . . . . . . . . . . . . 41

      A.5 Berkeley Unix dependencies . . . . . . . . . . . . . . . . . . 41

   B  Compatibility with past mistakes                                   43

      B.1 Living without a framing `type' byte . . . . . . . . . . . . . 43

      B.2 Backwards compatible SLIP servers  . . . . . . . . . . . . . . 43

   C  More aggressive compression                                        45

   D  Security Considerations                                            46

   E  Author's address                                                   46

   RFC 1144               Compressing TCP/IP Headers          February 1990

   1  Introduction

   As increasingly powerful computers find their way into people's homes,
   there is growing interest in extending Internet connectivity to those
   computers.  Unfortunately, this extension exposes some complex problems
   in link-level framing, address assignment, routing, authentication and
   performance.  As of this writing there is active work in all these
   areas.  This memo describes a method that has been used to improve
   TCP/IP performance over low speed (300 to 19,200 bps) serial links.

   The compression proposed here is similar in spirit to the Thinwire-II
   protocol described in [5].  However, this protocol compresses more
   effectively (the average compressed header is 3 bytes compared to 13 in
   Thinwire-II) and is both efficient and simple to implement (the Unix
   implementation is 250 lines of C and requires, on the average, 90us (170
   instructions) for a 20MHz MC68020 to compress or decompress a packet).

   This compression is specific to TCP/IP datagrams./2/  The author
   investigated compressing UDP/IP datagrams but found that they were too
   infrequent to be worth the bother and either there was insufficient
   datagram-to-datagram coherence for good compression (e.g., name server
   queries) or the higher level protocol headers overwhelmed the cost of
   the UDP/IP header (e.g., Sun's RPC/NFS). Separately compressing the IP
   and the TCP portions of the datagram was also investigated but rejected
   since it increased the average compressed header size by 50% and doubled
   the compression and decompression code size.

   2  The problem

   Internet services one might wish to access over a serial IP link from
   home range from interactive `terminal' type connections (e.g., telnet,
   rlogin, xterm) to bulk data transfer (e.g., ftp, smtp, nntp).  Header
   compression is motivated by the need for good interactive response.
   I.e., the line efficiency of a protocol is the ratio of the data to
   header+data in a datagram.  If efficient bulk data transfer is the only
   objective, it is always possible to make the datagram large enough to
   approach an efficiency of 100%.

   Human-factors studies[15] have found that interactive response is
   perceived as `bad' when low-level feedback (character echo) takes longer

     2. The tie to TCP is deeper than might be obvious.  In addition to the
   compression `knowing' the format of TCP and IP headers, certain features
   of TCP have been used to simplify the compression protocol.  In
   particular, TCP's reliable delivery and the byte-stream conversation
   model have been used to eliminate the need for any kind of error
   correction dialog in the protocol (see sec. 4).

   RFC 1144               Compressing TCP/IP Headers          February 1990

   than 100 to 200 ms.  Protocol headers interact with this threshold three

   (1) If the line is too slow, it may be impossible to fit both the
       headers and data into a 200 ms window:  One typed character results
       in a 41 byte TCP/IP packet being sent and a 41 byte echo being
       received.  The line speed must be at least 4000 bps to handle these
       82 bytes in 200 ms.

   (2) Even with a line fast enough to handle packetized typing echo (4800
       bps or above), there may be an undesirable interaction between bulk
       data and interactive traffic:  For reasonable line efficiency the
       bulk data packet size needs to be 10 to 20 times the header size.
       I.e., the line maximum transmission unit or MTU should be 500 to
       1000 bytes for 40 byte TCP/IP headers.  Even with type-of-service
       queuing to give priority to interactive traffic, a telnet packet has
       to wait for any in-progress bulk data packet to finish.  Assuming
       data transfer in only one direction, that wait averages half the MTU
       or 500 ms for a 1024 byte MTU at 9600 bps.

   (3) Any communication medium has a maximum signalling rate, the Shannon
       limit.  Based on an AT&T study[2], the Shannon limit for a typical
       dialup phone line is around 22,000 bps.  Since a full duplex, 9600
       bps modem already runs at 80% of the limit, modem manufacturers are
       starting to offer asymmetric allocation schemes to increase
       effective bandwidth:  Since a line rarely has equivalent amounts of
       data flowing both directions simultaneously, it is possible to give
       one end of the line more than 11,000 bps by either time-division
       multiplexing a half-duplex line (e.g., the Telebit Trailblazer) or
       offering a low-speed `reverse channel' (e.g., the USR Courier
       HST)./3/ In either case, the modem dynamically tries to guess which
       end of the conversation needs high bandwidth by assuming one end of
       the conversation is a human (i.e., demand is limited to <300 bps by
       typing speed).  The factor-of-forty bandwidth multiplication due to
       protocol headers will fool this allocation heuristic and cause these
       modems to `thrash'.

   From the above, it's clear that one design goal of the compression
   should be to limit the bandwidth demand of typing and ack traffic to at
   most 300 bps.  A typical maximum typing speed is around five characters

     3. See the excellent discussion of two-wire dialup line capacity in
   [1], chap. 11.  In particular, there is widespread misunderstanding of
   the capabilities of `echo-cancelling' modems (such as those conforming
   to CCITT V.32):  Echo-cancellation can offer each side of a two-wire
   line the full line bandwidth but, since the far talker's signal adds to
   the local `noise', not the full line capacity.  The 22Kbps Shannon limit
   is a hard-limit on data rate through a two-wire telephone connection.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   per second/4/ which leaves a budget 30 - 5 = 25 characters for headers
   or five bytes of header per character typed./5/  Five byte headers solve
   problems (1) and (3) directly and, indirectly, problem (2):  A packet
   size of 100--200 bytes will easily amortize the cost of a five byte
   header and offer a user 95--98% of the line bandwidth for data.  These
   short packets mean little interference between interactive and bulk data
   traffic (see sec. 5.2).

   Another design goal is that the compression protocol be based solely on
   information guaranteed to be known to both ends of a single serial link.
   Consider the topology shown in fig. 1 where communicating hosts A and B
   are on separate local area nets (the heavy black lines) and the nets are
   connected by two serial links (the open lines between gateways C--D and
   E--F)./6/ One compression possibility would be to convert each TCP/IP
   conversation into a semantically equivalent conversation in a protocol
   with smaller headers, e.g., to an X.25 call.  But, because of routing
   transients or multipathing, it's entirely possible that some of the A--B
   traffic will follow the A-C-D-B path and some will follow the A-E-F-B
   path.  Similarly, it's possible that A->B traffic will flow A-C-D-B and
   B->A traffic will flow B-F-E-A. None of the gateways can count on seeing
   all the packets in a particular TCP conversation and a compression
   algorithm that works for such a topology cannot be tied to the TCP
   connection syntax.

   A physical link treated as two, independent, simplex links (one each
   direction) imposes the minimum requirements on topology, routing and
   pipelining.  The ends of each simplex link only have to agree on the
   most recent packet(s) sent on that link.  Thus, although any compression
   scheme involves shared state, this state is spatially and temporally

     4. See [13].  Typing bursts or multiple character keystrokes such as
   cursor keys can exceed this average rate by factors of two to four.
   However the bandwidth demand stays approximately constant since the TCP
   Nagle algorithm[8] aggregates traffic with a <200ms interarrival time
   and the improved header-to-data ratio compensates for the increased
     5. A similar analysis leads to essentially the same header size limit
   for bulk data transfer ack packets.  Assuming that the MTU has been
   selected for `unobtrusive' background file transfers (i.e., chosen so
   the packet time is 200--400 ms --- see sec. 5), there can be at most 5
   data packets per second in the `high bandwidth' direction.  A reasonable
   TCP implementation will ack at most every other data packet so at 5
   bytes per ack the reverse channel bandwidth is 2.5 * 5 = 12.5 bytes/sec.
     6. Note that although the TCP endpoints are A and B, in this example
   compression/decompression must be done at the gateway serial links,
   i.e., between C and D and between E and F. Since A and B are using IP,
   they cannot know that their communication path includes a low speed
   serial link.  It is clearly a requirement that compression not break the
   IP model, i.e., that compression function between intermediate systems
   and not just between end systems.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   local and adheres to Dave Clark's principle of fate sharing[4]:  The two
   ends can only disagree on the state if the link connecting them is
   inoperable, in which case the disagreement doesn't matter.

   3  The compression algorithm

   3.1  The basic idea

   Figure 2 shows a typical (and minimum length) TCP/IP datagram header./7/
   The header size is 40 bytes:  20 bytes of IP and 20 of TCP.
   Unfortunately, since the TCP and IP protocols were not designed by a
   committee, all these header fields serve some useful purpose and it's
   not possible to simply omit some in the name of efficiency.

   However, TCP establishes connections and, typically, tens or hundreds of
   packets are exchanged on each connection.  How much of the per-packet
   information is likely to stay constant over the life of a connection?
   Half---the shaded fields in fig. 3.  So, if the sender and receiver keep
   track of active connections/8/ and the receiver keeps a copy of the
   header from the last packet it saw from each connection, the sender gets
   a factor-of-two compression by sending only a small (<= 8 bit)
   connection identifier together with the 20 bytes that change and letting
   the receiver fill in the 20 fixed bytes from the saved header.

   One can scavenge a few more bytes by noting that any reasonable
   link-level framing protocol will tell the receiver the length of a
   received message so total length (bytes 2 and 3) is redundant.  But then
   the header checksum (bytes 10 and 11), which protects individual hops
   from processing a corrupted IP header, is essentially the only part of
   the IP header being sent.  It seems rather silly to protect the
   transmission of information that isn't being transmitted.  So, the
   receiver can check the header checksum when the header is actually sent
   (i.e., in an uncompressed datagram) but, for compressed datagrams,
   regenerate it locally at the same time the rest of the IP header is
   being regenerated./9/

     7. The TCP and IP protocols and protocol headers are described in [10]
   and [11].
     8. The 96-bit tuple <src address, dst address, src port, dst port>
   uniquely identifies a TCP connection.
     9. The IP header checksum is not an end-to-end checksum in the sense
   of [14]:  The time-to-live update forces the IP checksum to be
   recomputed at each hop.  The author has had unpleasant personal
   experience with the consequences of violating the end-to-end argument in
   [14] and this protocol is careful to pass the end-to-end TCP checksum
   through unmodified.  See sec. 4.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   This leaves 16 bytes of header information to send.  All of these bytes
   are likely to change over the life of the conversation but they do not
   all change at the same time.  For example, during an FTP data transfer
   only the packet ID, sequence number and checksum change in the
   sender->receiver direction and only the packet ID, ack, checksum and,
   possibly, window, change in the receiver->sender direction.  With a copy
   of the last packet sent for each connection, the sender can figure out
   what fields change in the current packet then send a bitmask indicating
   what changed followed by the changing fields./10/

   If the sender only sends fields that differ, the above scheme gets the
   average header size down to around ten bytes.  However, it's worthwhile
   looking at how the fields change:  The packet ID typically comes from a
   counter that is incremented by one for each packet sent.  I.e., the
   difference between the current and previous packet IDs should be a
   small, positive integer, usually <256 (one byte) and frequently = 1.
   For packets from the sender side of a data transfer, the sequence number
   in the current packet will be the sequence number in the previous packet
   plus the amount of data in the previous packet (assuming the packets are
   arriving in order).  Since IP packets can be at most 64K, the sequence
   number change must be < 2^16 (two bytes).  So, if the differences in the
   changing fields are sent rather than the fields themselves, another
   three or four bytes per packet can be saved.

   That gets us to the five-byte header target.  Recognizing a couple of
   special cases will get us three byte headers for the two most common
   cases---interactive typing traffic and bulk data transfer---but the
   basic compression scheme is the differential coding developed above.
   Given that this intellectual exercise suggests it is possible to get
   five byte headers, it seems reasonable to flesh out the missing details
   and actually implement something.

   3.2  The ugly details

   3.2.1  Overview

   Figure 4 shows a block diagram of the compression software.  The
   networking system calls a SLIP output driver with an IP packet to be

    10. This is approximately Thinwire-I from [5].  A slight modification
   is to do a `delta encoding' where the sender subtracts the previous
   packet from the current packet (treating each packet as an array of 16
   bit integers), then sends a 20-bit mask indicating the non-zero
   differences followed by those differences.  If distinct conversations
   are separated, this is a fairly effective compression scheme (e.g.,
   typically 12-16 byte headers) that doesn't involve the compressor
   knowing any details of the packet structure.  Variations on this theme
   have been used, successfully, for a number of years (e.g., the Proteon
   router's serial link protocol[3]).

   RFC 1144               Compressing TCP/IP Headers          February 1990

   sent over the serial line.  The packet goes through a compressor which
   checks if the protocol is TCP. Non-TCP packets and `uncompressible' TCP
   packets (described below) are just marked as TYPE_IP and passed to a
   framer.  Compressible TCP packets are looked up in an array of packet
   headers.  If a matching connection is found, the incoming packet is
   compressed, the (uncompressed) packet header is copied into the array,
   and a packet of type COMPRESSED_TCP is sent to the framer.  If no match
   is found, the oldest entry in the array is discarded, the packet header
   is copied into that slot, and a packet of type UNCOMPRESSED_TCP is sent
   to the framer.  (An UNCOMPRESSED_TCP packet is identical to the original
   IP packet except the IP protocol field is replaced with a connection
   number---an index into the array of saved, per-connection packet
   headers.  This is how the sender (re-)synchronizes the receiver and
   `seeds' it with the first, uncompressed packet of a compressed packet

   The framer is responsible for communicating the packet data, type and
   boundary (so the decompressor can learn how many bytes came out of the
   compressor).  Since the compression is a differential coding, the framer
   must not re-order packets (this is rarely a concern over a single serial
   link).  It must also provide good error detection and, if connection
   numbers are compressed, must provide an error indication to the
   decompressor (see sec. 4)./11/

   The decompressor does a `switch' on the type of incoming packets:  For
   TYPE_IP, the packet is simply passed through.  For UNCOMPRESSED_TCP, the
   connection number is extracted from the IP protocol field and
   IPPROTO_TCP is restored, then the connection number is used as an index
   into the receiver's array of saved TCP/IP headers and the header of the
   incoming packet is copied into the indexed slot.  For COMPRESSED_TCP,
   the connection number is used as an array index to get the TCP/IP header
   of the last packet from that connection, the info in the compressed
   packet is used to update that header, then a new packet is constructed
   containing the now-current header from the array concatenated with the
   data from the compressed packet.

   Note that the communication is simplex---no information flows in the
   decompressor-to-compressor direction.  In particular, this implies that
   the decompressor is relying on TCP retransmissions to correct the saved
   state in the event of line errors (see sec. 4).

    11. Link level framing is outside the scope of this document.  Any
   framing that provides the facilities listed in this paragraph should be
   adequate for the compression protocol.  However, the author encourages
   potential implementors to see [9] for a proposed, standard, SLIP

   RFC 1144               Compressing TCP/IP Headers          February 1990

   3.2.2  Compressed packet format

   Figure 5 shows the format of a compressed TCP/IP packet.  There is a
   change mask that identifies which of the fields expected to change
   per-packet actually changed, a connection number so the receiver can
   locate the saved copy of the last packet for this TCP connection, the
   unmodified TCP checksum so the end-to-end data integrity check will
   still be valid, then for each bit set in the change mask, the amount the
   associated field changed.  (Optional fields, controlled by the mask, are
   enclosed in dashed lines in the figure.)  In all cases, the bit is set
   if the associated field is present and clear if the field is absent./12/

   Since the delta's in the sequence number, etc., are usually small,
   particularly if the tuning guidelines in section 5 are followed, all the
   numbers are encoded in a variable length scheme that, in practice,
   handles most traffic with eight bits:  A change of one through 255 is
   represented in one byte.  Zero is improbable (a change of zero is never
   sent) so a byte of zero signals an extension:  The next two bytes are
   the MSB and LSB, respectively, of a 16 bit value.  Numbers larger than
   16 bits force an uncompressed packet to be sent.  For example, decimal
   15 is encoded as hex 0f, 255 as ff, 65534 as 00 ff fe, and zero as 00 00
   00.  This scheme packs and decodes fairly efficiently:  The usual case
   for both encode and decode executes three instructions on a MC680x0.

   The numbers sent for TCP sequence number and ack are the difference/13/
   between the current value and the value in the previous packet (an
   uncompressed packet is sent if the difference is negative or more than
   64K). The number sent for the window is also the difference between the
   current and previous values.  However, either positive or negative
   changes are allowed since the window is a 16 bit field.  The packet's
   urgent pointer is sent if URG is set (an uncompressed packet is sent if
   the urgent pointer changes but URG is not set).  For packet ID, the
   number sent is the difference between the current and previous values.
   However, unlike the rest of the compressed fields, the assumed change
   when I is clear is one, not zero.

   There are two important special cases:

   (1) The sequence number and ack both change by the amount of data in the
       last packet; no window change or URG.

   (2) The sequence number changes by the amount of data in the last
       packet, no ack or window change or URG.

    12. The bit `P' in the figure is different from the others:  It is a
   copy of the `PUSH' bit from the TCP header.  `PUSH' is a curious
   anachronism considered indispensable by certain members of the Internet
   community.  Since PUSH can (and does) change in any datagram, an
   information preserving compression scheme must pass it explicitly.
    13. All differences are computed using two's complement arithmetic.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   (1) is the case for echoed terminal traffic.  (2) is the sender side of
   non-echoed terminal traffic or a unidirectional data transfer.  Certain
   combinations of the S, A, W and U bits of the change mask are used to
   signal these special cases.  `U' (urgent data) is rare so two unlikely
   combinations are S W U (used for case 1) and S A W U (used for case 2).
   To avoid ambiguity, an uncompressed packet is sent if the actual changes
   in a packet are S * W U.

   Since the `active' connection changes rarely (e.g., a user will type for
   several minutes in a telnet window before changing to a different
   window), the C bit allows the connection number to be elided.  If C is
   clear, the connection is assumed to be the same as for the last
   compressed or uncompressed packet.  If C is set, the connection number
   is in the byte immediately following the change mask./14/

   From the above, it's probably obvious that compressed terminal traffic
   usually looks like (in hex):  0B c c d, where the 0B indicates case (1),
   c c is the two byte TCP checksum and d is the character typed.  Commands
   to vi or emacs, or packets in the data transfer direction of an FTP
   `put' or `get' look like 0F c c d ... , and acks for that FTP look like
   04 c c a where a is the amount of data being acked./15/

   3.2.3  Compressor processing

   The compressor is called with the IP packet to be processed and the
   compression state structure for the outgoing serial line.  It returns a
   packet ready for final framing and the link level `type' of that packet.

   As the last section noted, the compressor converts every input packet
   into either a TYPE_IP, UNCOMPRESSED_TCP or COMPRESSED_TCP packet.  A

    14. The connection number is limited to one byte, i.e., 256
   simultaneously active TCP connections.  In almost two years of
   operation, the author has never seen a case where more than sixteen
   connection states would be useful (even in one case where the SLIP link
   was used as a gateway behind a very busy, 64-port terminal multiplexor).
   Thus this does not seem to be a significant restriction and allows the
   protocol field in UNCOMPRESSED_TCP packets to be used for the connection
   number, simplifying the processing of those packets.
    15. It's also obvious that the change mask changes infrequently and
   could often be elided.  In fact, one can do slightly better by saving
   the last compressed packet (it can be at most 16 bytes so this isn't
   much additional state) and checking to see if any of it (except the TCP
   checksum) has changed.  If not, send a packet type that means
   `compressed TCP, same as last time' and a packet containing only the
   checksum and data.  But, since the improvement is at most 25%, the added
   complexity and state doesn't seem justified.  See appendix C.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   TYPE_IP packet is an unmodified copy/16/ of the input packet and
   processing it doesn't change the compressor's state in any way.

   An UNCOMPRESSED_TCP packet is identical to the input packet except the
   IP protocol field (byte 9) is changed from `6' (protocol TCP) to a
   connection number.  In addition, the state slot associated with the
   connection number is updated with a copy of the input packet's IP and
   TCP headers and the connection number is recorded as the last connection
   sent on this serial line (for the C compression described below).

   A COMPRESSED_TCP packet contains the data, if any, from the original
   packet but the IP and TCP headers are completely replaced with a new,
   compressed header.  The connection state slot and last connection sent
   are updated by the input packet exactly as for an UNCOMPRESSED_TCP

   The compressor's decision procedure is:

     - If the packet is not protocol TCP, send it as TYPE_IP.

     - If the packet is an IP fragment (i.e., either the fragment offset
       field is non-zero or the more fragments bit is set), send it as

     - If any of the TCP control bits SYN, FIN or RST are set or if the ACK
       bit is clear, consider the packet uncompressible and send it as

    16. It is not necessary (or desirable) to actually duplicate the input
   packet for any of the three output types.  Note that the compressor
   cannot increase the size of a datagram.  As the code in appendix A
   shows, the protocol can be implemented so all header modifications are
   made `in place'.
    17. Only the first fragment contains the TCP header so the fragment
   offset check is necessary.  The first fragment might contain a complete
   TCP header and, thus, could be compressed.  However the check for a
   complete TCP header adds quite a lot of code and, given the arguments in
   [6], it seems reasonable to send all IP fragments uncompressed.
    18. The ACK test is redundant since a standard conforming
   implementation must set ACK in all packets except for the initial SYN
   packet.  However, the test costs nothing and avoids turning a bogus
   packet into a valid one.
   SYN packets are not compressed because only half of them contain a valid
   ACK field and they usually contain a TCP option (the max. segment size)
   which the following packets don't.  Thus the next packet would be sent
   uncompressed because the TCP header length changed and sending the SYN
   as UNCOMPRESSED_TCP instead of TYPE_IP would buy nothing.
   The decision to not compress FIN packets is questionable.  Discounting
   the trick in appendix B.1, there is a free bit in the header that could
   be used to communicate the FIN flag.  However, since connections tend to

   RFC 1144               Compressing TCP/IP Headers          February 1990

   If a packet makes it through the above checks, it will be sent as either

     - If no connection state can be found that matches the packet's source
       and destination IP addresses and TCP ports, some state is reclaimed
       (which should probably be the least recently used) and an
       UNCOMPRESSED_TCP packet is sent.

     - If a connection state is found, the packet header it contains is
       checked against the current packet to make sure there were no
       unexpected changes.  (E.g., that all the shaded fields in fig. 3 are
       the same).  The IP protocol, fragment offset, more fragments, SYN,
       FIN and RST fields were checked above and the source and destination
       address and ports were checked as part of locating the state.  So
       the remaining fields to check are protocol version, header length,
       type of service, don't fragment, time-to-live, data offset, IP
       options (if any) and TCP options (if any).  If any of these fields
       differ between the two headers, an UNCOMPRESSED_TCP packet is sent.

   If all the `unchanging' fields match, an attempt is made to compress the
   current packet:

     - If the URG flag is set, the urgent data field is encoded (note that
       it may be zero) and the U bit is set in the change mask.
       Unfortunately, if URG is clear, the urgent data field must be
       checked against the previous packet and, if it changes, an
       UNCOMPRESSED_TCP packet is sent.  (`Urgent data' shouldn't change
       when URG is clear but [11] doesn't require this.)

     - The difference between the current and previous packet's window
       field is computed and, if non-zero, is encoded and the W bit is set
       in the change mask.

     - The difference between ack fields is computed.  If the result is
       less than zero or greater than 2^16 - 1, an UNCOMPRESSED_TCP packet
       is sent./19/  Otherwise, if the result is non-zero, it is encoded
       and the A bit is set in the change mask.

     - The difference between sequence number fields is computed.  If the
       result is less than zero or greater than 2^16 - 1, an

   last for many packets, it seemed unreasonable to dedicate an entire bit
   to a flag that would only appear once in the lifetime of the connection.
    19. The two tests can be combined into a single test of the most
   significant 16 bits of the difference being non-zero.

   RFC 1144               Compressing TCP/IP Headers          February 1990

       UNCOMPRESSED_TCP packet is sent./20/  Otherwise, if the result is
       non-zero, it is encoded and the S bit is set in the change mask.

   Once the U, W, A and S changes have been determined, the special-case
   encodings can be checked:

     - If U, S and W are set, the changes match one of the special-case
       encodings.  Send an UNCOMPRESSED_TCP packet.

     - If only S is set, check if the change equals the amount of user data
       in the last packet.  I.e., subtract the TCP and IP header lengths
       from the last packet's total length field and compare the result to
       the S change.  If they're the same, set the change mask to SAWU (the
       special case for `unidirectional data transfer') and discard the
       encoded sequence number change (the decompressor can reconstruct it
       since it knows the last packet's total length and header length).

     - If only S and A are set, check if they both changed by the same
       amount and that amount is the amount of user data in the last
       packet.  If so, set the change mask to SWU (the special case for
       `echoed interactive' traffic) and discard the encoded changes.

     - If nothing changed, check if this packet has no user data (in which
       case it is probably a duplicate ack or window probe) or if the
       previous packet contained user data (which means this packet is a
       retransmission on a connection with no pipelining).  In either of
       these cases, send an UNCOMPRESSED_TCP packet.

   Finally, the TCP/IP header on the outgoing packet is replaced with a
   compressed header:

     - The change in the packet ID is computed and, if not one,/21/ the
       difference is encoded (note that it may be zero or negative) and the
       I bit is set in the change mask.

     - If the PUSH bit is set in the original datagram, the P bit is set in
       the change mask.

     - The TCP and IP headers of the packet are copied to the connection
       state slot.

    20. A negative sequence number change probably indicates a
   retransmission.  Since this may be due to the decompressor having
   dropped a packet, an uncompressed packet is sent to re-sync the
   decompressor (see sec. 4).
    21. Note that the test here is against one, not zero.  The packet ID is
   typically incremented by one for each packet sent so a change of zero is
   very unlikely.  A change of one is likely:  It occurs during any period
   when the originating system has activity on only one connection.

   RFC 1144               Compressing TCP/IP Headers          February 1990

     - The TCP and IP headers of the packet are discarded and a new header
       is prepended consisting of (in reverse order):

         - the accumulated, encoded changes.

         - the TCP checksum (if the new header is being constructed `in
           place', the checksum may have been overwritten and will have to
           be taken from the header copy in the connection state or saved
           in a temporary before the original header is discarded).

         - the connection number (if different than the last one sent on
           this serial line).  This also means that the the line's last
           connection sent must be set to the connection number and the C
           bit set in the change mask.

         - the change mask.

   At this point, the compressed TCP packet is passed to the framer for

   3.2.4  Decompressor processing

   Because of the simplex communication model, processing at the
   decompressor is much simpler than at the compressor --- all the
   decisions have been made and the decompressor simply does what the
   compressor has told it to do.

   The decompressor is called with the incoming packet,/22/ the length and
   type of the packet and the compression state structure for the incoming
   serial line.  A (possibly re-constructed) IP packet will be returned.

   The decompressor can receive four types of packet:  the three generated
   by the compressor and a TYPE_ERROR pseudo-packet generated when the
   receive framer detects an error./23/  The first step is a `switch' on
   the packet type:

     - If the packet is TYPE_ERROR or an unrecognized type, a `toss' flag
       is set in the state to force COMPRESSED_TCP packets to be discarded
       until one with the C bit set or an UNCOMPRESSED_TCP packet arrives.
       Nothing (a null packet) is returned.

    22. It's assumed that link-level framing has been removed by this point
   and the packet and length do not include type or framing bytes.
    23. No data need be associated with a TYPE_ERROR packet.  It exists so
   the receive framer can tell the decompressor that there may be a gap in
   the data stream.  The decompressor uses this as a signal that packets
   should be tossed until one arrives with an explicit connection number (C
   bit set).  See the last part of sec. 4.1 for a discussion of why this is

   RFC 1144               Compressing TCP/IP Headers          February 1990

     - If the packet is TYPE_IP, an unmodified copy of it is returned and
       the state is not modified.

     - If the packet is UNCOMPRESSED_TCP, the state index from the IP
       protocol field is checked./24/  If it's illegal, the toss flag is
       set and nothing is returned.  Otherwise, the toss flag is cleared,
       the index is copied to the state's last connection received field, a
       copy of the input packet is made,/25/ the TCP protocol number is
       restored to the IP protocol field, the packet header is copied to
       the indicated state slot, then the packet copy is returned.

   If the packet was not handled above, it is COMPRESSED_TCP and a new
   TCP/IP header has to be synthesized from information in the packet plus
   the last packet's header in the state slot.  First, the explicit or
   implicit connection number is used to locate the state slot:

     - If the C bit is set in the change mask, the state index is checked.
       If it's illegal, the toss flag is set and nothing is returned.
       Otherwise, last connection received is set to the packet's state
       index and the toss flag is cleared.

     - If the C bit is clear and the toss flag is set, the packet is
       ignored and nothing is returned.

   At this point, last connection received is the index of the appropriate
   state slot and the first byte(s) of the compressed packet (the change
   mask and, possibly, connection index) have been consumed.  Since the
   TCP/IP header in the state slot must end up reflecting the newly arrived
   packet, it's simplest to apply the changes from the packet to that
   header then construct the output packet from that header concatenated
   with the data from the input packet.  (In the following description,
   `saved header' is used as an abbreviation for `the TCP/IP header saved
   in the state slot'.)

     - The next two bytes in the incoming packet are the TCP checksum.
       They are copied to the saved header.

     - If the P bit is set in the change mask, the TCP PUSH bit is set in
       the saved header.  Otherwise the PUSH bit is cleared.

    24. State indices follow the C language convention and run from 0 to N
   - 1, where 0 < N <= 256 is the number of available state slots.
    25. As with the compressor, the code can be structured so no copies are
   done and all modifications are done in-place.  However, since the output
   packet can be larger than the input packet, 128 bytes of free space must
   be left at the front of the input packet buffer to allow room to prepend
   the TCP/IP header.

   RFC 1144               Compressing TCP/IP Headers          February 1990

     - If the low order four bits (S, A, W and U) of the change mask are
       all set (the `unidirectional data' special case), the amount of user
       data in the last packet is calculated by subtracting the TCP and IP
       header lengths from the IP total length in the saved header.  That
       amount is then added to the TCP sequence number in the saved header.

     - If S, W and U are set and A is clear (the `terminal traffic' special
       case), the amount of user data in the last packet is calculated and
       added to both the TCP sequence number and ack fields in the saved

     - Otherwise, the change mask bits are interpreted individually in the
       order that the compressor set them:

         - If the U bit is set, the TCP URG bit is set in the saved header
           and the next byte(s) of the incoming packet are decoded and
           stuffed into the TCP Urgent Pointer.  If the U bit is clear, the
           TCP URG bit is cleared.

         - If the W bit is set, the next byte(s) of the incoming packet are
           decoded and added to the TCP window field of the saved header.

         - If the A bit is set, the next byte(s) of the incoming packet are
           decoded and added to the TCP ack field of the saved header.

         - If the S bit is set, the next byte(s) of the incoming packet are
           decoded and added to the TCP sequence number field of the saved

     - If the I bit is set in the change mask, the next byte(s) of the
       incoming packet are decoded and added to the IP ID field of the
       saved packet.  Otherwise, one is added to the IP ID.

   At this point, all the header information from the incoming packet has
   been consumed and only data remains.  The length of the remaining data
   is added to the length of the saved IP and TCP headers and the result is
   put into the saved IP total length field.  The saved IP header is now up
   to date so its checksum is recalculated and stored in the IP checksum
   field.  Finally, an output datagram consisting of the saved header
   concatenated with the remaining incoming data is constructed and

   4  Error handling

   4.1  Error detection

   In the author's experience, dialup connections are particularly prone to
   data errors.  These errors interact with compression in two different

   RFC 1144               Compressing TCP/IP Headers          February 1990

   First is the local effect of an error in a compressed packet.  All error
   detection is based on redundancy yet compression has squeezed out almost
   all the redundancy in the TCP and IP headers.  In other words, the
   decompressor will happily turn random line noise into a perfectly valid
   TCP/IP packet./26/  One could rely on the TCP checksum to detect
   corrupted compressed packets but, unfortunately, some rather likely
   errors will not be detected.  For example, the TCP checksum will often
   not detect two single bit errors separated by 16 bits.  For a V.32 modem
   signalling at 2400 baud with 4 bits/baud, any line hit lasting longer
   than 400us. would corrupt 16 bits.  According to [2], residential phone
   line hits of up to 2ms. are likely.

   The correct way to deal with this problem is to provide for error
   detection at the framing level.  Since the framing (at least in theory)
   can be tailored to the characteristics of a particular link, the
   detection can be as light or heavy-weight as appropriate for that
   link./27/  Since packet error detection is done at the framing level,
   the decompressor simply assumes that it will get an indication that the
   current packet was received with errors.  (The decompressor always
   ignores (discards) a packet with errors.  However, the indication is
   needed to prevent the error being propagated --- see below.)

   The `discard erroneous packets' policy gives rise to the second
   interaction of errors and compression.  Consider the following

                 |original | sent   |received |reconstructed |
                 | 1:  A   | 1:  A  | 1:  A   | 1:  A        |
                 | 2:  BC  | 1,  BC | 1,  BC  | 2:  BC       |
                 | 4:  DE  | 2,  DE |  ---    |  ---         |
                 | 6:  F   | 2,  F  | 2,  F   | 4:  F        |
                 | 7:  GH  | 1,  GH | 1,  GH  | 5:  GH       |

   (Each entry above has the form `starting sequence number:data sent' or
   `?sequence number change,data sent'.)  The first thing sent is an
   uncompressed packet, followed by four compressed packets.  The third
   packet picks up an error and is discarded.  To reconstruct the fourth
   packet, the receiver applies the sequence number change from incoming
   compressed packet to the sequence number of the last correctly received

    26. modulo the TCP checksum.
    27. While appropriate error detection is link dependent, the CCITT CRC
   used in [9] strikes an excellent balance between ease of computation and
   robust error detection for a large variety of links, particularly at the
   relatively small packet sizes needed for good interactive response.
   Thus, for the sake of interoperability, the framing in [9] should be
   used unless there is a truly compelling reason to do otherwise.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   packet, packet two, and generates an incorrect sequence number for
   packet four.  After the error, all reconstructed packets' sequence
   numbers will be in error, shifted down by the amount of data in the
   missing packet./28/

   Without some sort of check, the preceding error would result in the
   receiver invisibly losing two bytes from the middle of the transfer
   (since the decompressor regenerates sequence numbers, the packets
   containing F and GH arrive at the receiver's TCP with exactly the
   sequence numbers they would have had if the DE packet had never
   existed).  Although some TCP conversations can survive missing data/29/
   it is not a practice to be encouraged.  Fortunately the TCP checksum,
   since it is a simple sum of the packet contents including the sequence
   numbers, detects 100% of these errors.  E.g., the receiver's computed
   checksum for the last two packets above always differs from the packet
   checksum by two.

   Unfortunately, there is a way for the TCP checksum protection described
   above to fail if the changes in an incoming compressed packet are
   applied to the wrong conversation:  Consider two active conversations C1
   and C2 and a packet from C1 followed by two packets from C2.  Since the
   connection number doesn't change, it's omitted from the second C2
   packet.  But, if the first C2 packet is received with a CRC error, the
   second C2 packet will mistakenly be considered the next packet in C1.
   Since the C2 checksum is a random number with respect to the C1 sequence
   numbers, there is at least a 2^-16 probability that this packet will be
   accepted by the C1 TCP receiver./30/  To prevent this, after a CRC error
   indication from the framer the receiver discards packets until it
   receives either a COMPRESSED_TCP packet with the C bit set or an
   UNCOMPRESSED_TCP packet.  I.e., packets are discarded until the receiver
   gets an explicit connection number.

   To summarize this section, there are two different types of errors:
   per-packet corruption and per-conversation loss-of-sync.  The first type
   is detected at the decompressor from a link-level CRC error, the second
   at the TCP receiver from a (guaranteed) invalid TCP checksum.  The
   combination of these two independent mechanisms ensures that erroneous
   packets are discarded.

    28. This is an example of a generic problem with differential or delta
   encodings known as `losing DC'.
    29. Many system managers claim that holes in an NNTP stream are more
   valuable than the data.
    30. With worst-case traffic, this probability translates to one
   undetected error every three hours over a 9600 baud line with a 30%
   error rate).

   RFC 1144               Compressing TCP/IP Headers          February 1990

   4.2  Error recovery

   The previous section noted that after a CRC error the decompressor will
   introduce TCP checksum errors in every uncompressed packet.  Although
   the checksum errors prevent data stream corruption, the TCP conversation
   won't be terribly useful until the decompressor again generates valid
   packets.  How can this be forced to happen?

   The decompressor generates invalid packets because its state (the saved
   `last packet header') disagrees with the compressor's state.  An
   UNCOMPRESSED_TCP packet will correct the decompressor's state.  Thus
   error recovery amounts to forcing an uncompressed packet out of the
   compressor whenever the decompressor is (or might be) confused.

   The first thought is to take advantage of the full duplex communication
   link and have the decompressor send something to the compressor
   requesting an uncompressed packet.  This is clearly undesirable since it
   constrains the topology more than the minimum suggested in sec. 2 and
   requires that a great deal of protocol be added to both the decompressor
   and compressor.  A little thought convinces one that this alternative is
   not only undesirable, it simply won't work:  Compressed packets are
   small and it's likely that a line hit will so completely obliterate one
   that the decompressor will get nothing at all.  Thus packets are
   reconstructed incorrectly (because of the missing compressed packet) but
   only the TCP end points, not the decompressor, know that the packets are

   But the TCP end points know about the error and TCP is a reliable
   protocol designed to run over unreliable media.  This means the end
   points must eventually take some sort of error recovery action and
   there's an obvious trigger for the compressor to resync the
   decompressor:  send uncompressed packets whenever TCP is doing error

   But how does the compressor recognize TCP error recovery?  Consider the
   schematic TCP data transfer of fig. 6.    The confused decompressor is
   in the forward (data transfer) half of the TCP conversation.  The
   receiving TCP discards packets rather than acking them (because of the
   checksum errors), the sending TCP eventually times out and retransmits a
   packet, and the forward path compressor finds that the difference
   between the sequence number in the retransmitted packet and the sequence
   number in the last packet seen is either negative (if there were
   multiple packets in transit) or zero (one packet in transit).  The first
   case is detected in the compression step that computes sequence number
   differences.  The second case is detected in the step that checks the
   `special case' encodings but needs an additional test:  It's fairly
   common for an interactive conversation to send a dataless ack packet
   followed by a data packet.  The ack and data packet will have the same
   sequence numbers yet the data packet is not a retransmission.  To
   prevent sending an unnecessary uncompressed packet, the length of the
   previous packet should be checked and, if it contained data, a zero

   RFC 1144               Compressing TCP/IP Headers          February 1990

   sequence number change must indicate a retransmission.

   A confused decompressor in the reverse (ack) half of the conversation is
   as easy to detect (fig. 7):    The sending TCP discards acks (because
   they contain checksum errors), eventually times out, then retransmits
   some packet.  The receiving TCP thus gets a duplicate packet and must
   generate an ack for the next expected sequence number[11, p. 69].  This
   ack will be a duplicate of the last ack the receiver generated so the
   reverse-path compressor will find no ack, seq number, window or urg
   change.  If this happens for a packet that contains no data, the
   compressor assumes it is a duplicate ack sent in response to a
   retransmit and sends an UNCOMPRESSED_TCP packet./31/

   5  Configurable parameters and tuning

   5.1  Compression configuration

   There are two configuration parameters associated with header
   compression:  Whether or not compressed packets should be sent on a
   particular line and, if so, how many state slots (saved packet headers)
   to reserve.  There is also one link-level configuration parameter, the
   maximum packet size or MTU, and one front-end configuration parameter,
   data compression, that interact with header compression.  Compression
   configuration is discussed in this section.  MTU and data compression
   are discussed in the next two sections.

   There are some hosts (e.g., low end PCs) which may not have enough
   processor or memory resources to implement this compression.  There are
   also rare link or application characteristics that make header
   compression unnecessary or undesirable.  And there are many existing
   SLIP links that do not currently use this style of header compression.
   For the sake of interoperability, serial line IP drivers that allow
   header compression should include some sort of user configurable flag to
   disable compression (see appendix B.2)./32/

   If compression is enabled, the compressor must be sure to never send a
   connection id (state index) that will be dropped by the decompressor.
   E.g., a black hole is created if the decompressor has sixteen slots and

    31. The packet could be a zero-window probe rather than a retransmitted
   ack but window probes should be infrequent and it does no harm to send
   them uncompressed.
    32. The PPP protocol in [9] allows the end points to negotiate
   compression so there is no interoperability problem.  However, there
   should still be a provision for the system manager at each end to
   control whether compression is negotiated on or off.  And, obviously,
   compression should default to `off' until it has been negotiated `on'.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   the compressor uses twenty./33/  Also, if the compressor is allowed too
   few slots, the LRU allocator will thrash and most packets will be sent
   as UNCOMPRESSED_TCP. Too many slots and memory is wasted.

   Experimenting with different sizes over the past year, the author has
   found that eight slots will thrash (i.e., the performance degradation is
   noticeable) when many windows on a multi-window workstation are
   simultaneously in use or the workstation is being used as a gateway for
   three or more other machines.  Sixteen slots were never observed to
   thrash.  (This may simply be because a 9600 bps line split more than 16
   ways is already so overloaded that the additional degradation from
   round-robbining slots is negligible.)

   Each slot must be large enough to hold a maximum length TCP/IP header of
   128 bytes/34/ so 16 slots occupy 2KB of memory.  In these days of 4 Mbit
   RAM chips, 2KB seems so little memory that the author recommends the
   following configuration rules:

   (1) If the framing protocol does not allow negotiation, the compressor
       and decompressor should provide sixteen slots, zero through fifteen.

   (2) If the framing protocol allows negotiation, any mutually agreeable
       number of slots from 1 to 256 should be negotiable./35/  If number
       of slots is not negotiated, or until it is negotiated, both sides
       should assume sixteen.

   (3) If you have complete control of all the machines at both ends of
       every link and none of them will ever be used to talk to machines
       outside of your control, you are free to configure them however you
       please, ignoring the above.  However, when your little eastern-block
       dictatorship collapses (as they all eventually seem to), be aware
       that a large, vocal, and not particularly forgiving Internet
       community will take great delight in pointing out to anyone willing

    33. Strictly speaking, there's no reason why the connection id should
   be treated as an array index.  If the decompressor's states were kept in
   a hash table or other associative structure, the connection id would be
   a key, not an index, and performance with too few decompressor slots
   would only degrade enormously rather than failing altogether.  However,
   an associative structure is substantially more costly in code and cpu
   time and, given the small per-slot cost (128 bytes of memory), it seems
   reasonable to design for slot arrays at the decompressor and some
   (possibly implicit) communication of the array size.
    34. The maximum header length, fixed by the protocol design, is 64
   bytes of IP and 64 bytes of TCP.
    35. Allowing only one slot may make the compressor code more complex.
   Implementations should avoid offering one slot if possible and
   compressor implementations may disable compression if only one slot is

   RFC 1144               Compressing TCP/IP Headers          February 1990

       to listen that you have misconfigured your systems and are not

   5.2  Choosing a maximum transmission unit

   From the discussion in sec. 2, it seems desirable to limit the maximum
   packet size (MTU) on any line where there might be interactive traffic
   and multiple active connections (to maintain good interactive response
   between the different connections competing for the line).  The obvious
   question is `how much does this hurt throughput?'  It doesn't.

   Figure 8 shows how user data throughput/36/ scales with MTU with (solid
   line) and without (dashed line) header compression.  The dotted lines
   show what MTU corresponds to a 200 ms packet time at 2400, 9600 and
   19,200 bps.  Note that with header compression even a 2400 bps line can
   be responsive yet have reasonable throughput (83%)./37/

   Figure 9 shows how line efficiency scales with increasing line speed,
   assuming that a 200ms. MTU is always chosen./38/  The knee in the
   performance curve is around 2400 bps.  Below this, efficiency is
   sensitive to small changes in speed (or MTU since the two are linearly
   related) and good efficiency comes at the expense of good response.
   Above 2400bps the curve is flat and efficiency is relatively independent
   of speed or MTU. In other words, it is possible to have both good
   response and high line efficiency.

   To illustrate, note that for a 9600 bps line with header compression
   there is essentially no benefit in increasing the MTU beyond 200 bytes:
   If the MTU is increased to 576, the average delay increases by 188%
   while throughput only improves by 3% (from 96 to 99%).

    36. The vertical axis is in percent of line speed.  E.g., `95' means
   that 95% of the line bandwidth is going to user data or, in other words,
   the user would see a data transfer rate of 9120 bps on a 9600 bps line.
   Four bytes of link-level (framer) encapsulation in addition to the
   TCP/IP or compressed header were included when calculating the relative
   throughput.  The 200 ms packet times were computed assuming an
   asynchronous line using 10 bits per character (8 data bits, 1 start, 1
   stop, no parity).
    37. However, the 40 byte TCP MSS required for a 2400 bps line might
   stress-test your TCP implementation.
    38. For a typical async line, a 200ms. MTU is simply .02 times the line
   speed in bits per second.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   5.3  Interaction with data compression

   Since the early 1980's, fast, effective, data compression algorithms
   such as Lempel-Ziv[7] and programs that embody them, such as the
   compress program shipped with Berkeley Unix, have become widely
   available.  When using low speed or long haul lines, it has become
   common practice to compress data before sending it.  For dialup
   connections, this compression is often done in the modems, independent
   of the communicating hosts.  Some interesting issues would seem to be:
   (1) Given a good data compressor, is there any need for header
   compression?  (2) Does header compression interact with data
   compression?  (3) Should data be compressed before or after header

   To investigate (1), Lempel-Ziv compression was done on a trace of 446
   TCP/IP packets taken from the user's side of a typical telnet
   conversation.  Since the packets resulted from typing, almost all
   contained only one data byte plus 40 bytes of header.  I.e., the test
   essentially measured L-Z compression of TCP/IP headers.  The compression
   ratio (the ratio of uncompressed to compressed data) was 2.6.  In other
   words, the average header was reduced from 40 to 16 bytes.  While this
   is good compression, it is far from the 5 bytes of header needed for
   good interactive response and far from the 3 bytes of header (a
   compression ratio of 13.3) that header compression yielded on the same
   packet trace.

   The second and third questions are more complex.  To investigate them,
   several packet traces from FTP file transfers were analyzed/40/ with and
   without header compression and with and without L-Z compression.  The
   L-Z compression was tried at two places in the outgoing data stream
   (fig. 10):    (1) just before the data was handed to TCP for
   encapsulation (simulating compression done at the `application' level)
   and (2) after the data was encapsulated (simulating compression done in
   the modem).  Table 1 summarizes the results for a 78,776 byte ASCII text
   file (the Unix csh.1 manual entry)/41/ transferred using the guidelines
   of the previous section (256 byte MTU or 216 byte MSS; 368 packets
   total).  Compression ratios for the following ten tests are shown
   (reading left to right and top to bottom):

    39. The answers, for those who wish to skip the remainder of this
   section, are `yes', `no' and `either', respectively.
    40. The data volume from user side of a telnet is too small to benefit
   from data compression and can be adversely affected by the delay most
   compression algorithms (necessarily) add.  The statistics and volume of
   the computer side of a telnet are similar to an (ASCII) FTP so these
   results should apply to either.
    41. The ten experiments described were each done on ten ASCII files
   (four long e-mail messages, three Unix C source files and three Unix
   manual entries).  The results were remarkably similar for different
   files and the general conclusions reached below apply to all ten files.

   RFC 1144               Compressing TCP/IP Headers          February 1990

     - data file (no compression or encapsulation)

     - data -> L--Z compressor

     - data -> TCP/IP encapsulation

     - data -> L--Z -> TCP/IP

     - data -> TCP/IP -> L--Z

     - data -> L--Z -> TCP/IP -> L--Z

     - data -> TCP/IP -> Hdr. Compress.

     - data -> L--Z -> TCP/IP -> Hdr. Compress.

     - data -> TCP/IP -> Hdr. Compress. -> L--Z

     - data -> L--Z -> TCP/IP -> Hdr. Compress. -> L--Z

            |              | No data  | L--Z   |  L--Z  |  L--Z   |
            |              |compress. |on data |on wire | on both |
            | Raw Data     |     1.00 |   2.44 |   ---- |    ---- |
            | + TCP Encap. |     0.83 |   2.03 |   1.97 |    1.58 |
            | w/Hdr Comp.  |     0.98 |   2.39 |   2.26 |    1.66 |

                 Table 1:  ASCII Text File Compression Ratios

   The first column of table 1 says the data expands by 19% (`compresses'
   by .83) when encapsulated in TCP/IP and by 2% when encapsulated in
   header compressed TCP/IP./42/ The first row says L--Z compression is
   quite effective on this data, shrinking it to less than half its
   original size.  Column four illustrates the well-known fact that it is a
   mistake to L--Z compress already compressed data.  The interesting
   information is in rows two and three of columns two and three.  These
   columns say that the benefit of data compression overwhelms the cost of
   encapsulation, even for straight TCP/IP. They also say that it is
   slightly better to compress the data before encapsulating it rather than
   compressing at the framing/modem level.  The differences however are

    42. This is what would be expected from the relative header sizes:
   256/216 for TCP/IP and 219/216 for header compression.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   small --- 3% and 6%, respectively, for the TCP/IP and header compressed

   Table 2 shows the same experiment for a 122,880 byte binary file (the
   Sun-3 ps executable).  Although the raw data doesn't compress nearly as
   well, the results are qualitatively the same as for the ASCII data.  The
   one significant change is in row two:  It is about 3% better to compress
   the data in the modem rather than at the source if doing TCP/IP
   encapsulation (apparently, Sun binaries and TCP/IP headers have similar
   statistics).  However, with header compression (row three) the results
   were similar to the ASCII data --- it's about 3% worse to compress at
   the modem rather than the source./44/

            |              | No data  | L--Z   |  L--Z  |  L--Z   |
            |              |compress. |on data |on wire | on both |
            | Raw Data     |     1.00 |   1.72 |   ---- |    ---- |
            | + TCP Encap. |     0.83 |   1.43 |   1.48 |    1.21 |
            | w/Hdr Comp.  |     0.98 |   1.69 |   1.64 |    1.28 |

                   Table 2:  Binary File Compression Ratios

   6  Performance measurements

   An implementation goal of compression code was to arrive at something
   simple enough to run at ISDN speeds (64Kbps) on a typical 1989

    43. The differences are due to the wildly different byte patterns of
   TCP/IP datagrams and ASCII text.  Any compression scheme with an
   underlying, Markov source model, such as Lempel-Ziv, will do worse when
   radically different sources are interleaved.  If the relative
   proportions of the two sources are changed, i.e., the MTU is increased,
   the performance difference between the two compressor locations
   decreases.  However, the rate of decrease is very slow --- increasing
   the MTU by 400% (256 to 1024) only changed the difference between the
   data and modem L--Z choices from 2.5% to 1.3%.
    44. There are other good reasons to compress at the source:  Far fewer
   packets have to be encapsulated and far fewer characters have to be sent
   to the modem.  The author suspects that the `compress data in the modem'
   alternative should be avoided except when faced with an intractable,
   vendor proprietary operating system.

   RFC 1144               Compressing TCP/IP Headers          February 1990

                   |               |  Average per-packet   |
                   |    Machine    | processing time (us.) |
                   |               |                       |
                   |               | Compress | Decompress |
                   |Sparcstation-1 |       24 |         18 |
                   |   Sun 4/260   |       46 |         20 |
                   |   Sun 3/60    |       90 |         90 |
                   |   Sun 3/50    |      130 |        150 |
                   |  HP9000/370   |       42 |         33 |
                   |  HP9000/360   |       68 |         70 |
                   |   DEC 3100    |       27 |         25 |
                   |    Vax 780    |      430 |        300 |
                   |    Vax 750    |      800 |        500 |
                   |   CCI Tahoe   |      110 |        140 |

                      Table 3:  Compression code timings

   workstation.  64Kbps is a byte every 122us so 120us was (arbitrarily)
   picked as the target compression/decompression time./45/

   As part of the compression code development, a trace-driven exerciser
   was developed.  This was initially used to compare different compression
   protocol choices then later to test the code on different computer
   architectures and do regression tests after performance `improvements'.
   A small modification of this test program resulted in a useful
   measurement tool./46/  Table 3 shows the result of timing the
   compression code on all the machines available to the author (times were
   measured using a mixed telnet/ftp traffic trace).  With the exception of
   the Vax architectures, which suffer from (a) having bytes in the wrong
   order and (b) a lousy compiler (Unix pcc), all machines essentially met
   the 120us goal.

    45. The time choice wasn't completely arbitrary:  Decompression is
   often done during the inter-frame `flag' character time so, on systems
   where the decompression is done at the same priority level as the serial
   line input interrupt, times much longer than a character time would
   result in receiver overruns.  And, with the current average of five byte
   frames (on the wire, including both compressed header and framing), a
   compression/decompression that takes one byte time can use at most 20%
   of the available time.  This seems like a comfortable budget.
    46. Both the test program and timer program are included in the
   ftp-able package described in appendix A as files tester.c and timer.c.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   7  Acknowlegements

   The author is grateful to the members of the Internet Engineering Task
   Force, chaired by Phill Gross, who provided encouragement and thoughtful
   review of this work.  Several patient beta-testers, particularly Sam
   Leffler and Craig Leres, tracked down and fixed problems in the initial
   implementation.  Cynthia Livingston and Craig Partridge carefully read
   and greatly improved an unending sequence of partial drafts of this
   document.  And last but not least, Telebit modem corporation,
   particularly Mike Ballard, encouraged this work from its inception and
   has been an ongoing champion of serial line and dial-up IP.


    [1] Bingham, J. A. C. Theory and Practice of Modem Design. John Wiley
        & Sons, 1988.

    [2] Carey, M. B., Chan, H.-T., Descloux, A., Ingle, J. F., and Park,
        K. I. 1982/83 end office connection study:  Analog voice and
        voiceband data transmission performance characterization of the
        public switched network. Bell System Technical Journal 63, 9 (Nov.

    [3] Chiappa, N., 1988. Private communication.

    [4] Clark, D. D. The design philosophy of the DARPA Internet
        protocols. In Proceedings of SIGCOMM '88 (Stanford, CA, Aug.
        1988), ACM.

    [5] Farber, D. J., Delp, G. S., and Conte, T. M. A Thinwire Protocol
        for connecting personal computers to the Internet. Arpanet Working
        Group Requests for Comment, DDN Network Information Center, SRI
        International, Menlo Park, CA, Sept. 1984. RFC-914.

    [6] Kent, C. A., and Mogul, J. Fragmentation considered harmful. In
        Proceedings of SIGCOMM '87 (Aug. 1987), ACM.

    [7] Lempel, A., and Ziv, J. Compression of individual sequences via
        variable-rate encoding. IEEE Transactions on Information Theory
        IT-24, 5 (June 1978).

    [8] Nagle, J. Congestion Control in IP/TCP Internetworks. Arpanet
        Working Group Requests for Comment, DDN Network Information Center,
        SRI International, Menlo Park, CA, Jan. 1984. RFC-896.

    [9] Perkins, D. Point-to-Point Protocol:  A proposal for
        multi-protocol transmission of datagrams over point-to-point links.
        Arpanet Working Group Requests for Comment, DDN Network Information
        Center, SRI International, Menlo Park, CA, Nov. 1989. RFC-1134.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   [10] Postel, J., Ed. Internet Protocol Specification. SRI
        International, Menlo Park, CA, Sept. 1981. RFC-791.

   [11] Postel, J., Ed. Transmission Control Protocol Specification. SRI
        International, Menlo Park, CA, Sept. 1981. RFC-793.

   [12] Romkey, J. A Nonstandard for Transmission of IP Datagrams Over
        Serial Lines:  Slip. Arpanet Working Group Requests for Comment,
        DDN Network Information Center, SRI International, Menlo Park, CA,
        June 1988. RFC-1055.

   [13] Salthouse, T. A. The skill of typing. Scientific American 250, 2
        (Feb. 1984), 128--135.

   [14] Saltzer, J. H., Reed, D. P., and Clark, D. D. End-to-end arguments
        in system design. ACM Transactions on Computer Systems 2, 4 (Nov.

   [15] Shneiderman, B. Designing the User Interface. Addison-Wesley,

   RFC 1144               Compressing TCP/IP Headers          February 1990

   A  Sample Implementation

   The following is a sample implementation of the protocol described in
   this document.

   Since many people who might have the deal with this code are familiar
   with the Berkeley Unix kernel and its coding style (affectionately known
   as kernel normal form), this code was done in that style.  It uses the
   Berkeley `subroutines' (actually, macros and/or inline assembler
   expansions) for converting to/from network byte order and
   copying/comparing strings of bytes.  These routines are briefly
   described in sec. A.5 for anyone not familiar with them.

   This code has been run on all the machines listed in the table on page
   24.  Thus, the author hopes there are no byte order or alignment
   problems (although there are embedded assumptions about alignment that
   are valid for Berkeley Unix but may not be true for other IP
   implementations --- see the comments mentioning alignment in
   sl_compress_tcp and sl_decompress_tcp).

   There was some attempt to make this code efficient.  Unfortunately, that
   may have made portions of it incomprehensible.  The author apologizes
   for any frustration this engenders.  (In honesty, my C style is known to
   be obscure and claims of `efficiency' are simply a convenient excuse.)

   This sample code and a complete Berkeley Unix implementation is
   available in machine readable form via anonymous ftp from Internet host
   ftp.ee.lbl.gov (, file cslip.tar.Z. This is a compressed
   Unix tar file.  It must be ftped in binary mode.

   All of the code in this appendix is covered by the following copyright:

    * Copyright (c) 1989 Regents of the University of California.
    * All rights reserved.
    * Redistribution and use in source and binary forms are
    * permitted provided that the above copyright notice and this
    * paragraph are duplicated in all such forms and that any
    * documentation, advertising materials, and other materials
    * related to such distribution and use acknowledge that the
    * software was developed by the University of California,
    * Berkeley.  The name of the University may not be used to
    * endorse or promote products derived from this software
    * without specific prior written permission.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   A.1  Definitions and State Data

   #define MAX_STATES 16   /* must be >2 and <255 */
   #define MAX_HDR 128     /* max TCP+IP hdr length (by protocol def) */

   /* packet types */
   #define TYPE_IP 0x40
   #define TYPE_COMPRESSED_TCP 0x80
   #define TYPE_ERROR 0x00 /* this is not a type that ever appears on
                            * the wire.  The receive framer uses it to
                            * tell the decompressor there was a packet
                            * transmission error. */
    * Bits in first octet of compressed packet

   /* flag bits for what changed in a packet */

   #define NEW_C  0x40
   #define NEW_I  0x20
   #define TCP_PUSH_BIT 0x10

   #define NEW_S  0x08
   #define NEW_A  0x04
   #define NEW_W  0x02
   #define NEW_U  0x01

   /* reserved, special-case values of above */
   #define SPECIAL_I (NEW_S|NEW_W|NEW_U)        /* echoed interactive traffic */
   #define SPECIAL_D (NEW_S|NEW_A|NEW_W|NEW_U)  /* unidirectional data */

    * "state" data for each active tcp conversation on the wire.  This is
    * basically a copy of the entire IP/TCP header from the last packet together
    * with a small identifier the transmit & receive ends of the line use to
    * locate saved header.
   struct cstate {
        struct cstate *cs_next;  /* next most recently used cstate (xmit only) */
        u_short cs_hlen;         /* size of hdr (receive only) */
        u_char cs_id;            /* connection # associated with this state */
        u_char cs_filler;
        union {
             char hdr[MAX_HDR];
             struct ip csu_ip;   /* ip/tcp hdr from most recent packet */
        } slcs_u;
   #define cs_ip slcs_u.csu_ip

   RFC 1144               Compressing TCP/IP Headers          February 1990

   #define cs_hdr slcs_u.csu_hdr

    * all the state data for one serial line (we need one of these per line).
   struct slcompress {
        struct cstate *last_cs;            /* most recently used tstate */
        u_char last_recv;                  /* last rcvd conn. id */
        u_char last_xmit;                  /* last sent conn. id */
        u_short flags;
        struct cstate tstate[MAX_STATES];  /* xmit connection states */
        struct cstate rstate[MAX_STATES];  /* receive connection states */

   /* flag values */
   #define SLF_TOSS 1       /* tossing rcvd frames because of input err */

    * The following macros are used to encode and decode numbers.  They all
    * assume that `cp' points to a buffer where the next byte encoded (decoded)
    * is to be stored (retrieved).  Since the decode routines do arithmetic,
    * they have to convert from and to network byte order.

    * ENCODE encodes a number that is known to be non-zero.  ENCODEZ checks for
    * zero (zero has to be encoded in the long, 3 byte form).
   #define ENCODE(n) { \
        if ((u_short)(n) >= 256) { \
             *cp++ = 0; \
             cp[1] = (n); \
             cp[0] = (n) >> 8; \
             cp += 2; \
        } else { \
             *cp++ = (n); \
        } \
   #define ENCODEZ(n) { \
        if ((u_short)(n) >= 256 || (u_short)(n) == 0) { \
             *cp++ = 0; \
             cp[1] = (n); \
             cp[0] = (n) >> 8; \
             cp += 2; \
        } else { \
             *cp++ = (n); \
        } \

    * DECODEL takes the (compressed) change at byte cp and adds it to the

   RFC 1144               Compressing TCP/IP Headers          February 1990

    * current value of packet field 'f' (which must be a 4-byte (long) integer
    * in network byte order).  DECODES does the same for a 2-byte (short) field.
    * DECODEU takes the change at cp and stuffs it into the (short) field f.
    * 'cp' is updated to point to the next field in the compressed header.
   #define DECODEL(f) { \
        if (*cp == 0) {\
             (f) = htonl(ntohl(f) + ((cp[1] << 8) | cp[2])); \
             cp += 3; \
        } else { \
             (f) = htonl(ntohl(f) + (u_long)*cp++); \
        } \
   #define DECODES(f) { \
        if (*cp == 0) {\
             (f) = htons(ntohs(f) + ((cp[1] << 8) | cp[2])); \
             cp += 3; \
        } else { \
             (f) = htons(ntohs(f) + (u_long)*cp++); \
        } \
   #define DECODEU(f) { \
        if (*cp == 0) {\
             (f) = htons((cp[1] << 8) | cp[2]); \
             cp += 3; \
        } else { \
             (f) = htons((u_long)*cp++); \
        } \

   RFC 1144               Compressing TCP/IP Headers          February 1990

   A.2  Compression

   This routine looks daunting but isn't really.  The code splits into four
   approximately equal sized sections:  The first quarter manages a
   circularly linked, least-recently-used list of `active' TCP
   connections./47/  The second figures out the sequence/ack/window/urg
   changes and builds the bulk of the compressed packet.  The third handles
   the special-case encodings.  The last quarter does packet ID and
   connection ID encoding and replaces the original packet header with the
   compressed header.

   The arguments to this routine are a pointer to a packet to be
   compressed, a pointer to the compression state data for the serial line,
   and a flag which enables or disables connection id (C bit) compression.

   Compression is done `in-place' so, if a compressed packet is created,
   both the start address and length of the incoming packet (the off and
   len fields of m) will be updated to reflect the removal of the original
   header and its replacement by the compressed header.  If either a
   compressed or uncompressed packet is created, the compression state is
   updated.  This routines returns the packet type for the transmit framer

   Because 16 and 32 bit arithmetic is done on various header fields, the
   incoming IP packet must be aligned appropriately (e.g., on a SPARC, the
   IP header is aligned on a 32-bit boundary).  Substantial changes would
   have to be made to the code below if this were not true (and it would
   probably be cheaper to byte copy the incoming header to somewhere
   correctly aligned than to make those changes).

   Note that the outgoing packet will be aligned arbitrarily (e.g., it
   could easily start on an odd-byte boundary).

   sl_compress_tcp(m, comp, compress_cid)
        struct mbuf *m;
        struct slcompress *comp;
        int compress_cid;
        register struct cstate *cs = comp->last_cs->cs_next;
        register struct ip *ip = mtod(m, struct ip *);
        register u_int hlen = ip->ip_hl;
        register struct tcphdr *oth;       /* last TCP header */
        register struct tcphdr *th;        /* current TCP header */

    47. The two most common operations on the connection list are a `find'
   that terminates at the first entry (a new packet for the most recently
   used connection) and moving the last entry on the list to the head of
   the list (the first packet from a new connection).  A circular list
   efficiently handles these two operations.

   RFC 1144               Compressing TCP/IP Headers          February 1990

        register u_int deltaS, deltaA;     /* general purpose temporaries */
        register u_int changes = 0;        /* change mask */
        u_char new_seq[16];                /* changes from last to current */
        register u_char *cp = new_seq;

         * Bail if this is an IP fragment or if the TCP packet isn't
         * `compressible' (i.e., ACK isn't set or some other control bit is
         * set).  (We assume that the caller has already made sure the packet
         * is IP proto TCP).
        if ((ip->ip_off & htons(0x3fff)) || m->m_len < 40)
             return (TYPE_IP);

        th = (struct tcphdr *) & ((int *) ip)[hlen];
        if ((th->th_flags & (TH_SYN | TH_FIN | TH_RST | TH_ACK)) != TH_ACK)
             return (TYPE_IP);

         * Packet is compressible -- we're going to send either a
         * COMPRESSED_TCP or UNCOMPRESSED_TCP packet.  Either way we need to
         * locate (or create) the connection state.  Special case the most
         * recently used connection since it's most likely to be used again &
         * we don't have to do any reordering if it's used.
        if (ip->ip_src.s_addr != cs->cs_ip.ip_src.s_addr ||
            ip->ip_dst.s_addr != cs->cs_ip.ip_dst.s_addr ||
            *(int *) th != ((int *) &cs->cs_ip)[cs->cs_ip.ip_hl]) {

              * Wasn't the first -- search for it.
              * States are kept in a circularly linked list with last_cs
              * pointing to the end of the list.  The list is kept in lru
              * order by moving a state to the head of the list whenever
              * it is referenced.  Since the list is short and,
              * empirically, the connection we want is almost always near
              * the front, we locate states via linear search.  If we
              * don't find a state for the datagram, the oldest state is
              * (re-)used.
             register struct cstate *lcs;
             register struct cstate *lastcs = comp->last_cs;

             do {
                  lcs = cs;
                  cs = cs->cs_next;
                  if (ip->ip_src.s_addr == cs->cs_ip.ip_src.s_addr
                      && ip->ip_dst.s_addr == cs->cs_ip.ip_dst.s_addr
                      && *(int *) th == ((int *) &cs->cs_ip)[cs->cs_ip.ip_hl])
                       goto found;

   RFC 1144               Compressing TCP/IP Headers          February 1990

             } while (cs != lastcs);

              * Didn't find it -- re-use oldest cstate.  Send an
              * uncompressed packet that tells the other side what
              * connection number we're using for this conversation. Note
              * that since the state list is circular, the oldest state
              * points to the newest and we only need to set last_cs to
              * update the lru linkage.
             comp->last_cs = lcs;
             hlen += th->th_off;
             hlen <<= 2;
             goto uncompressed;

             /* Found it -- move to the front on the connection list. */
             if (lastcs == cs)
                  comp->last_cs = lcs;
             else {
                  lcs->cs_next = cs->cs_next;
                  cs->cs_next = lastcs->cs_next;
                  lastcs->cs_next = cs;
         * Make sure that only what we expect to change changed. The first
         * line of the `if' checks the IP protocol version, header length &
         * type of service.  The 2nd line checks the "Don't fragment" bit.
         * The 3rd line checks the time-to-live and protocol (the protocol
         * check is unnecessary but costless).  The 4th line checks the TCP
         * header length.  The 5th line checks IP options, if any.  The 6th
         * line checks TCP options, if any.  If any of these things are
         * different between the previous & current datagram, we send the
         * current datagram `uncompressed'.
        oth = (struct tcphdr *) & ((int *) &cs->cs_ip)[hlen];
        deltaS = hlen;
        hlen += th->th_off;
        hlen <<= 2;

        if (((u_short *) ip)[0] != ((u_short *) &cs->cs_ip)[0] ||
            ((u_short *) ip)[3] != ((u_short *) &cs->cs_ip)[3] ||
            ((u_short *) ip)[4] != ((u_short *) &cs->cs_ip)[4] ||
            th->th_off != oth->th_off ||
            (deltaS > 5 && BCMP(ip + 1, &cs->cs_ip + 1, (deltaS - 5) << 2)) ||
            (th->th_off > 5 && BCMP(th + 1, oth + 1, (th->th_off - 5) << 2)))
             goto uncompressed;

         * Figure out which of the changing fields changed.  The receiver

   RFC 1144               Compressing TCP/IP Headers          February 1990

         * expects changes in the order: urgent, window, ack, seq.
        if (th->th_flags & TH_URG) {
             deltaS = ntohs(th->th_urp);
             changes |= NEW_U;
        } else if (th->th_urp != oth->th_urp)
              * argh! URG not set but urp changed -- a sensible
              * implementation should never do this but RFC793 doesn't
              * prohibit the change so we have to deal with it.
             goto uncompressed;

        if (deltaS = (u_short) (ntohs(th->th_win) - ntohs(oth->th_win))) {
             changes |= NEW_W;
        if (deltaA = ntohl(th->th_ack) - ntohl(oth->th_ack)) {
             if (deltaA > 0xffff)
                  goto uncompressed;
             changes |= NEW_A;
        if (deltaS = ntohl(th->th_seq) - ntohl(oth->th_seq)) {
             if (deltaS > 0xffff)
                  goto uncompressed;
             changes |= NEW_S;
         * Look for the special-case encodings.
        switch (changes) {

        case 0:
              * Nothing changed. If this packet contains data and the last
              * one didn't, this is probably a data packet following an
              * ack (normal on an interactive connection) and we send it
              * compressed.  Otherwise it's probably a retransmit,
              * retransmitted ack or window probe.  Send it uncompressed
              * in case the other side missed the compressed version.
             if (ip->ip_len != cs->cs_ip.ip_len &&
                 ntohs(cs->cs_ip.ip_len) == hlen)

             /* (fall through) */

        case SPECIAL_I:

   RFC 1144               Compressing TCP/IP Headers          February 1990

        case SPECIAL_D:
              * Actual changes match one of our special case encodings --
              * send packet uncompressed.
             goto uncompressed;

        case NEW_S | NEW_A:
             if (deltaS == deltaA &&
                 deltaS == ntohs(cs->cs_ip.ip_len) - hlen) {
                  /* special case for echoed terminal traffic */
                  changes = SPECIAL_I;
                  cp = new_seq;

        case NEW_S:
             if (deltaS == ntohs(cs->cs_ip.ip_len) - hlen) {
                  /* special case for data xfer */
                  changes = SPECIAL_D;
                  cp = new_seq;
        deltaS = ntohs(ip->ip_id) - ntohs(cs->cs_ip.ip_id);
        if (deltaS != 1) {
             changes |= NEW_I;
        if (th->th_flags & TH_PUSH)
             changes |= TCP_PUSH_BIT;
         * Grab the cksum before we overwrite it below.  Then update our
         * state with this packet's header.
        deltaA = ntohs(th->th_sum);
        BCOPY(ip, &cs->cs_ip, hlen);

         * We want to use the original packet as our compressed packet. (cp -
         * new_seq) is the number of bytes we need for compressed sequence
         * numbers.  In addition we need one byte for the change mask, one
         * for the connection id and two for the tcp checksum. So, (cp -
         * new_seq) + 4 bytes of header are needed.  hlen is how many bytes
         * of the original packet to toss so subtract the two to get the new
         * packet size.
        deltaS = cp - new_seq;
        cp = (u_char *) ip;
        if (compress_cid == 0 || comp->last_xmit != cs->cs_id) {
             comp->last_xmit = cs->cs_id;

   RFC 1144               Compressing TCP/IP Headers          February 1990

             hlen -= deltaS + 4;
             cp += hlen;
             *cp++ = changes | NEW_C;
             *cp++ = cs->cs_id;
        } else {
             hlen -= deltaS + 3;
             cp += hlen;
             *cp++ = changes;
        m->m_len -= hlen;
        m->m_off += hlen;
        *cp++ = deltaA >> 8;
        *cp++ = deltaA;
        BCOPY(new_seq, cp, deltaS);
        return (TYPE_COMPRESSED_TCP);

         * Update connection state cs & send uncompressed packet
         * ('uncompressed' means a regular ip/tcp packet but with the
         * 'conversation id' we hope to use on future compressed packets in
         * the protocol field).
        BCOPY(ip, &cs->cs_ip, hlen);
        ip->ip_p = cs->cs_id;
        comp->last_xmit = cs->cs_id;
        return (TYPE_UNCOMPRESSED_TCP);

   RFC 1144               Compressing TCP/IP Headers          February 1990

   A.3  Decompression

   This routine decompresses a received packet.  It is called with a
   pointer to the packet, the packet length and type, and a pointer to the
   compression state structure for the incoming serial line.  It returns a
   pointer to the resulting packet or zero if there were errors in the
   incoming packet.  If the packet is COMPRESSED_TCP or UNCOMPRESSED_TCP,
   the compression state will be updated.

   The new packet will be constructed in-place.  That means that there must
   be 128 bytes of free space in front of bufp to allow room for the
   reconstructed IP and TCP headers.  The reconstructed packet will be
   aligned on a 32-bit boundary.

   u_char *
   sl_uncompress_tcp(bufp, len, type, comp)
        u_char *bufp;
        int len;
        u_int type;
        struct slcompress *comp;
        register u_char *cp;
        register u_int hlen, changes;
        register struct tcphdr *th;
        register struct cstate *cs;
        register struct ip *ip;

        switch (type) {

        case TYPE_ERROR:
             goto bad;

        case TYPE_IP:
             return (bufp);

              * Locate the saved state for this connection.  If the state
              * index is legal, clear the 'discard' flag.
             ip = (struct ip *) bufp;
             if (ip->ip_p >= MAX_STATES)
                  goto bad;

             cs = &comp->rstate[comp->last_recv = ip->ip_p];
             comp->flags &= ~SLF_TOSS;
              * Restore the IP protocol field then save a copy of this
              * packet header.  (The checksum is zeroed in the copy so we
              * don't have to zero it each time we process a compressed

   RFC 1144               Compressing TCP/IP Headers          February 1990

              * packet.
             ip->ip_p = IPPROTO_TCP;
             hlen = ip->ip_hl;
             hlen += ((struct tcphdr *) & ((int *) ip)[hlen])->th_off;
             hlen <<= 2;
             BCOPY(ip, &cs->cs_ip, hlen);
             cs->cs_ip.ip_sum = 0;
             cs->cs_hlen = hlen;
             return (bufp);

        /* We've got a compressed packet. */
        cp = bufp;
        changes = *cp++;
        if (changes & NEW_C) {
              * Make sure the state index is in range, then grab the
              * state. If we have a good state index, clear the 'discard'
              * flag.
             if (*cp >= MAX_STATES)
                  goto bad;

             comp->flags &= ~SLF_TOSS;
             comp->last_recv = *cp++;
        } else {
              * This packet has an implicit state index.  If we've had a
              * line error since the last time we got an explicit state
              * index, we have to toss the packet.
             if (comp->flags & SLF_TOSS)
                  return ((u_char *) 0);
         * Find the state then fill in the TCP checksum and PUSH bit.
        cs = &comp->rstate[comp->last_recv];
        hlen = cs->cs_ip.ip_hl << 2;
        th = (struct tcphdr *) & ((u_char *) &cs->cs_ip)[hlen];
        th->th_sum = htons((*cp << 8) | cp[1]);
        cp += 2;
        if (changes & TCP_PUSH_BIT)
             th->th_flags |= TH_PUSH;
             th->th_flags &= ~TH_PUSH;


   RFC 1144               Compressing TCP/IP Headers          February 1990

         * Fix up the state's ack, seq, urg and win fields based on the
         * changemask.
        switch (changes & SPECIALS_MASK) {
        case SPECIAL_I:
             register u_int i = ntohs(cs->cs_ip.ip_len) - cs->cs_hlen;
             th->th_ack = htonl(ntohl(th->th_ack) + i);
             th->th_seq = htonl(ntohl(th->th_seq) + i);

        case SPECIAL_D:
             th->th_seq = htonl(ntohl(th->th_seq) + ntohs(cs->cs_ip.ip_len)
                          - cs->cs_hlen);

             if (changes & NEW_U) {
                  th->th_flags |= TH_URG;
             } else
                  th->th_flags &= ~TH_URG;
             if (changes & NEW_W)
             if (changes & NEW_A)
             if (changes & NEW_S)
        /* Update the IP ID */
        if (changes & NEW_I)
             cs->cs_ip.ip_id = htons(ntohs(cs->cs_ip.ip_id) + 1);

         * At this point, cp points to the first byte of data in the packet.
         * If we're not aligned on a 4-byte boundary, copy the data down so
         * the IP & TCP headers will be aligned.  Then back up cp by the
         * TCP/IP header length to make room for the reconstructed header (we
         * assume the packet we were handed has enough space to prepend 128
         * bytes of header).  Adjust the lenth to account for the new header
         * & fill in the IP total length.
        len -= (cp - bufp);
        if (len < 0)
              * we must have dropped some characters (crc should detect
              * this but the old slip framing won't)

   RFC 1144               Compressing TCP/IP Headers          February 1990

             goto bad;

        if ((int) cp & 3) {
             if (len > 0)
                  OVBCOPY(cp, (int) cp & ~3, len);
             cp = (u_char *) ((int) cp & ~3);
        cp -= cs->cs_hlen;
        len += cs->cs_hlen;
        cs->cs_ip.ip_len = htons(len);
        BCOPY(&cs->cs_ip, cp, cs->cs_hlen);

        /* recompute the ip header checksum */
             register u_short *bp = (u_short *) cp;
             for (changes = 0; hlen > 0; hlen -= 2)
                  changes += *bp++;
             changes = (changes & 0xffff) + (changes >> 16);
             changes = (changes & 0xffff) + (changes >> 16);
             ((struct ip *) cp)->ip_sum = ~changes;
        return (cp);

        comp->flags |= SLF_TOSS;
        return ((u_char *) 0);

   RFC 1144               Compressing TCP/IP Headers          February 1990

   A.4  Initialization

   This routine initializes the state structure for both the transmit and
   receive halves of some serial line.  It must be called each time the
   line is brought up.

        struct slcompress *comp;
        register u_int i;
        register struct cstate *tstate = comp->tstate;

         * Clean out any junk left from the last time line was used.
        bzero((char *) comp, sizeof(*comp));
         * Link the transmit states into a circular list.
        for (i = MAX_STATES - 1; i > 0; --i) {
             tstate[i].cs_id = i;
             tstate[i].cs_next = &tstate[i - 1];
        tstate[0].cs_next = &tstate[MAX_STATES - 1];
        tstate[0].cs_id = 0;
        comp->last_cs = &tstate[0];
         * Make sure we don't accidentally do CID compression
         * (assumes MAX_STATES < 255).
        comp->last_recv = 255;
        comp->last_xmit = 255;

   A.5  Berkeley Unix dependencies

   Note:  The following is of interest only if you are trying to bring the
   sample code up on a system that is not derived from 4BSD (Berkeley

   The code uses the normal Berkeley Unix header files (from
   /usr/include/netinet) for definitions of the structure of IP and TCP
   headers.  The structure tags tend to follow the protocol RFCs closely
   and should be obvious even if you do not have access to a 4BSD

    48. In the event they are not obvious, the header files (and all the
   Berkeley networking code) can be anonymous ftp'd from host

   RFC 1144               Compressing TCP/IP Headers          February 1990

   The macro BCOPY(src, dst, amt) is invoked to copy amt bytes from src to
   dst.  In BSD, it translates into a call to bcopy.  If you have the
   misfortune to be running System-V Unix, it can be translated into a call
   to memcpy.  The macro OVBCOPY(src, dst, amt) is used to copy when src
   and dst overlap (i.e., when doing the 4-byte alignment copy).  In the
   BSD kernel, it translates into a call to ovbcopy.  Since AT&T botched
   the definition of memcpy, this should probably translate into a copy
   loop under System-V.

   The macro BCMP(src, dst, amt) is invoked to compare amt bytes of src and
   dst for equality.  In BSD, it translates into a call to bcmp.  In
   System-V, it can be translated into a call to memcmp or you can write a
   routine to do the compare.  The routine should return zero if all bytes
   of src and dst are equal and non-zero otherwise.

   The routine ntohl(dat) converts (4 byte) long dat from network byte
   order to host byte order.  On a reasonable cpu this can be the no-op
                           #define ntohl(dat) (dat)

   On a Vax or IBM PC (or anything with Intel byte order), you will have to
   define a macro or routine to rearrange bytes.

   The routine ntohs(dat) is like ntohl but converts (2 byte) shorts
   instead of longs.  The routines htonl(dat) and htons(dat) do the inverse
   transform (host to network byte order) for longs and shorts.

   A struct mbuf is used in the call to sl_compress_tcp because that
   routine needs to modify both the start address and length if the
   incoming packet is compressed.  In BSD, an mbuf is the kernel's buffer
   management structure.  If other systems, the following definition should
   be sufficient:

            struct mbuf {
                    u_char  *m_off; /* pointer to start of data */
                    int     m_len;  /* length of data */

            #define mtod(m, t) ((t)(m->m_off))

   ucbarpa.berkeley.edu, files pub/4.3/tcp.tar and pub/4.3/inet.tar.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   B  Compatibility with past mistakes

   When combined with the modern PPP serial line protocol[9], the use of
   header compression is automatic and invisible to the user.
   Unfortunately, many sites have existing users of the SLIP described in
   [12] which doesn't allow for different protocol types to distinguish
   header compressed packets from IP packets or for version numbers or an
   option exchange that could be used to automatically negotiate header

   The author has used the following tricks to allow header compressed SLIP
   to interoperate with the existing servers and clients.  Note that these
   are hacks for compatibility with past mistakes and should be offensive
   to any right thinking person.  They are offered solely to ease the pain
   of running SLIP while users wait patiently for vendors to release PPP.

   B.1  Living without a framing `type' byte

   The bizarre packet type numbers in sec. A.1 were chosen to allow a
   `packet type' to be sent on lines where it is undesirable or impossible
   to add an explicit type byte.  Note that the first byte of an IP packet
   always contains `4' (the IP protocol version) in the top four bits.  And
   that the most significant bit of the first byte of the compressed header
   is ignored.  Using the packet types in sec. A.1, the type can be encoded
   in the most significant bits of the outgoing packet using the code

                    p->dat[0] |= sl_compress_tcp(p, comp);

    and decoded on the receive side by

                  if (p->dat[0] & 0x80)
                          type = TYPE_COMPRESSED_TCP;
                  else if (p->dat[0] >= 0x70) {
                          type = TYPE_UNCOMPRESSED_TCP;
                          p->dat[0] &=~ 0x30;
                  } else
                          type = TYPE_IP;
                  status = sl_uncompress_tcp(p, type, comp);

   B.2  Backwards compatible SLIP servers

   The SLIP described in [12] doesn't include any mechanism that could be
   used to automatically negotiate header compression.  It would be nice to

   RFC 1144               Compressing TCP/IP Headers          February 1990

   allow users of this SLIP to use header compression but, when users of
   the two SLIP varients share a common server, it would be annoying and
   difficult to manually configure both ends of each connection to enable
   compression.  The following procedure can be used to avoid manual

   Since there are two types of dial-in clients (those that implement
   compression and those that don't) but one server for both types, it's
   clear that the server will be reconfiguring for each new client session
   but clients change configuration seldom if ever.  If manual
   configuration has to be done, it should be done on the side that changes
   infrequently --- the client.  This suggests that the server should
   somehow learn from the client whether to use header compression.
   Assuming symmetry (i.e., if compression is used at all it should be used
   both directions) the server can use the receipt of a compressed packet
   from some client to indicate that it can send compressed packets to that
   client.  This leads to the following algorithm:

   There are two bits per line to control header compression:  allowed and
   on.  If on is set, compressed packets are sent, otherwise not.  If
   allowed is set, compressed packets can be received and, if an
   UNCOMPRESSED_TCP packet arrives when on is clear, on will be set./49/
   If a compressed packet arrives when allowed is clear, it will be

   Clients are configured with both bits set (allowed is always set if on
   is set) and the server starts each session with allowed set and on
   clear.  The first compressed packet from the client (which must be a
   UNCOMPRESSED_TCP packet) turns on compression for the server.

    49. Since [12] framing doesn't include error detection, one should be
   careful not to `false trigger' compression on the server.  The
   UNCOMPRESSED_TCP packet should checked for consistency (e.g., IP
   checksum correctness) before compression is enabled.  Arrival of
   COMPRESSED_TCP packets should not be used to enable compression.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   C  More aggressive compression

   As noted in sec. 3.2.2, easily detected patterns exist in the stream of
   compressed headers, indicating that more compression could be done.
   Would this be worthwhile?

   The average compressed datagram has only seven bits of header./50/  The
   framing must be at least one bit (to encode the `type') and will
   probably be more like two to three bytes.  In most interesting cases
   there will be at least one byte of data.  Finally, the end-to-end
   check---the TCP checksum---must be passed through unmodified./51/

   The framing, data and checksum will remain even if the header is
   completely compressed out so the change in average packet size is, at
   best, four bytes down to three bytes and one bit --- roughly a 25%
   improvement in delay./52/  While this may seem significant, on a 2400
   bps line it means that typing echo response takes 25 rather than 29 ms.
   At the present stage of human evolution, this difference is not

   However, the author sheepishly admits to perverting this compression
   scheme for a very special case data-acquisition problem:  We had an
   instrument and control package floating at 200KV, communicating with
   ground level via a telemetry system.  For many reasons (multiplexed
   communication, pipelining, error recovery, availability of well tested
   implementations, etc.), it was convenient to talk to the package using
   TCP/IP. However, since the primary use of the telemetry link was data
   acquisition, it was designed with an uplink channel capacity <0.5% the
   downlink's.  To meet application delay budgets, data packets were 100
   bytes and, since TCP acks every other packet, the relative uplink
   bandwidth for acks is a/200 where `a' is the total size of ack packets.
   Using the scheme in this paper, the smallest ack is four bytes which
   would imply an uplink bandwidth 2% of the downlink.  This wasn't

    50. Tests run with several million packets from a mixed traffic load
   (i.e., statistics kept on a year's traffic from my home to work) show
   that 80% of packets use one of the two special encodings and, thus, the
   only header is the change mask.
    51. If someone tries to sell you a scheme that compresses the TCP
   checksum `Just say no'.  Some poor fool has yet to have the sad
   experience that reveals the end-to-end argument is gospel truth.  Worse,
   since the fool is subverting your end-to-end error check, you may pay
   the price for this education and they will be none the wiser.  What does
   it profit a man to gain two byte times of delay and lose peace of mind?
    52. Note again that we must be concerned about interactive delay to be
   making this argument:  Bulk data transfer performance will be dominated
   by the time to send the data and the difference between three and four
   byte headers on a datagram containing tens or hundreds of data bytes is,
   practically, no difference.

   RFC 1144               Compressing TCP/IP Headers          February 1990

   possible so we used the scheme described in footnote 15:  If the first
   bit of the frame was one, it meant `same compressed header as last
   time'.  Otherwise the next two bits gave one of the types described in
   sec. 3.2.  Since the link had excellent forward error correction and
   traffic made only a single hop, the TCP checksum was compressed out
   (blush!) of the `same header' packet types/53/ so the total header size
   for these packets was one bit.  Over several months of operation, more
   than 99% of the 40 byte TCP/IP headers were compressed down to one

   D  Security Considerations

   Security considerations are not addressed in this memo.

   E  Author's address

       Address:  Van Jacobson
                 Real Time Systems Group
                 Mail Stop 46A
                 Lawrence Berkeley Laboratory
                 Berkeley, CA 94720

       Phone:    Use email (author ignores his phone)

       EMail:    van@helios.ee.lbl.gov

    53. The checksum was re-generated in the decompressor and, of course,
   the `toss' logic was made considerably more aggressive to prevent error
    54. We have heard the suggestion that `real-time' needs require
   abandoning TCP/IP in favor of a `light-weight' protocol with smaller
   headers.  It is difficult to envision a protocol that averages less than
   one header bit per packet.