Internet Engineering Task Force      Audio/Video Transport Working Group
INTERNET-DRAFT                              S. Casner / Precept Software
draft-ietf-avt-crtp-01.txt                            V. Jacobson / LBNL
                                                       November 25, 1996
                                                           Expires: 5/97


       Compressing IP/UDP/RTP Headers for Low-Speed Serial Links

Status of this Memo

This document is an Internet-Draft.  Internet-Drafts are working docu-
ments of the Internet Engineering Task Force (IETF), its areas, and its
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Distribution of this document is unlimited.

                                Abstract

     This document describes a method for compressing the headers of
     IP/UDP/RTP datagrams to reduce overhead on low-speed serial links.
     In many cases, all three headers can be compressed to 2-4 bytes.

Comments are solicited and should be addressed to the working group
mailing list rem-conf@es.net and/or the author(s).  This draft updates
draft-casner-jacobson-crtp-00.txt which was sent to the rem-conf list
but was never officially posted as an Internet-Draft due to a mail los-
sage that then left it out of date.  At the Montreal IETF meeting in
June 1996, this proposal was accepted as a work item of the the
Audio/Video Transport working group, hence the draft name change.









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1.  Introduction

Since the Real-time Transport Protocol was published as an RFC [1],
there has been growing interest in using RTP as one step to achieve
interoperability among different implementations of network audio/video
applications.  However, there is also concern that the 12-byte RTP
header is too large an overhead for 20-byte payloads when operating over
low speed lines such as dial-up modems at 14.4 or 28.8 kb/s.  (Existing
applications operating in this environment may use an application-
specific protocol with a header of a few bytes that has reduced func-
tionality relative to RTP.)

Header size may be reduced through compression techniques as has been
done with great success for TCP [2].  In this case, compression might be
applied to the RTP header alone, on an end-to-end basis, or to the com-
bination of IP, UDP and RTP headers on a link-by-link basis.  Compress-
ing the 40 bytes of combined headers together provides substantially
more gain than compressing 12 bytes of RTP header alone because the
resulting size is approximately the same (2-4 bytes) in either case.
Compressing on a link-by-link basis also provides better performance
because the delay and loss rate are lower.  Therefore, the method
defined here is for combined compression of IP, UDP and RTP headers on a
link-by-link basis.

This document defines a compression scheme that may be used with IPv4,
IPv6 or packets encapsulated with more than one IP header, though the
initial focus is on IPv4.  It is intended that the IP/UDP/RTP compres-
sion defined here will fit within and be referenced by the the more com-
plete compression framework [3] specified by Mikael Degermark, et. al.,
which covers both IPv6 and IPv4 with TCP and non-TCP as two classes of
transport above IP.  IP/UDP/RTP would be extracted as a third class from
the non-TCP class.

2.  Assumptions and Tradeoffs

The goal of this compression scheme is to reduce the IP/UDP/RTP headers
to two bytes for most packets in the case where no UDP checksums are
being sent, or four bytes with checksums.  It is motivated primarily by
the specific problem of sending audio and video over 14.4 and 28.8
dialup modems.  These links tend to provide full-duplex communication,
so the protocol takes advantage of that fact, though this constraint
could be removed.

This specification does not address segmentation and preemption of large
packets to reduce the delay across the slow link experienced by small
real-time packets, except to identify in Section 4 some interactions
between segmentation and compression that may occur.  Segmentation
schemes may be defined separately and used in conjunction with the



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compression defined here.

It should be noted that implementation simplicity is an important factor
to consider in evaluating the a compression scheme.  Communications
servers may need to support compression over perhaps as many as 100
dial-up modem lines using a single processor.  Therefore, it may be
appropriate to make some simplifications in the design at the expense of
generality, or to produce a flexible design that is general but can be
subsetted for simplicity.  The next sections discuss some of the trade-
offs listed here.

2.1.  Simplex vs. Full Duplex

In the absence of other constraints, a compression scheme that worked
over simplex links would be preferred over one that did not.  However,
operation over a simplex link requires periodic refreshes with an
uncompressed packet header to restore compression state in case of
error.  If an explicit error signal can be returned instead, the delay
to recovery may be shortened substantially.  The overhead in the no-
error case is also reduced.  Some UDP applications may require only sim-
plex communication, but RTP applications will frequently require full
duplex communication.  The application may be 2-way, as in a telephone
conversation, but even if data flows in only one direction there is a
need for a return path to carry reception feedback in RTCP packets.

This specification includes an error indication on the reverse path,
however it would be possible to use a periodic refresh instead.  When-
ever the decompressor detected an error in a particular packet stream,
it would simply discard all packets in that stream until an uncompressed
header for was received for that stream, and then resume decompression.
The penalty would be the potentially large number of packets discarded.

2.2.  Segmentation and Layering

Delay induced by the time required to send a large packet over the slow
link is not a problem for one-way audio, for example, because the
receiver can adapt to the variance in delay.  However, for interactive
conversations, minimizing the end-to-end delay is critical.  Segmenta-
tion of large, none-real-time packets to allow small real-time packets
to be transmitted between segments can reduce the delay.

This specification deals only with compression and assumes segmentation,
if included, will be handled as a separate layer.  It seems inappropri-
ate to integrate segmentation and compression in such a way that the
compression could not be used by itself in situations where segmentation
was deemed unnecessary or impractical.  Similarly, one would like to
avoid any requirements for a reservation protocol.  The compression
scheme can be applied locally on the two ends of a link independent of



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any other mechanisms except for the requirements that the link layer
provide some packet type codes, a packet length indication, and good
error detection.

Conversely, separately compressing the IP/UDP and RTP layers loses too
much of the compression gain that is possible by treating them together.
Crossing these protocol layer boundaries is appropriate because the same
function is being applied across all layers.

3.  The Compression Algorithm

The compression algorithm defined in this document draws heavily upon
the design of TCP/IP header compression as described in RFC 1144 [2].
Readers are referred to that RFC for more information on the underlying
motivations and general principles of header compression.

3.1.  The basic idea

In TCP header compression, the first factor of two comes from the obser-
vation that half of the bytes in the header remain constant over the
life of the connection.  After sending the uncompressed header once,
these fields may be elided from the compressed headers that follow.  The
remaining compression comes from differential coding on the changing
fields to reduce their size, and from eliminating the changing fields
entirely for common cases by calculating the changes from the length of
the packet.  This length is indicated by the link-level protocol.

For RTP header compression, some of the same techniques may be applied.
However, the big gain comes from the observation that although several
fields change in every packet, the difference from packet to packet is
often constant and therefore the second-order difference is zero.  By
maintaining both the uncompressed header and the first-order differences
in the session state shared between the compressor and decompressor, all
that must be communicated is an indication that the second-order differ-
ence was zero.  The decompressor can reconstruct the original header
without any loss of information.

Just as TCP/IP header compression maintains shared state for multiple
simultaneous TCP connections, this IP/UDP/RTP compression must maintain
state for multiple session contexts.  A session context is defined by
the combination of the IP source and destination addresses, the UDP
source and destination ports, and the RTP SSRC field.

Because the RTP compression is lossless, it may be applied to any UDP
traffic that benefits from it.  Most likely, the only packets that will
benefit are RTP packets, but it is acceptable to use heuristics to
determine whether or not the packet is an RTP packet because no harm is
done if the heurisic gives the wrong answer.  This does require



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executing the compression algorithm for all UDP packets.  Most implemen-
tations will need to maintain a negative cache of packet streams (iden-
tified by addresses and ports but not the SSRC field) that have failed
to compress as RTP packets for some number of attempts.  Failing to
compress means that the fields that are expected to remain constant most
of the time, such as the payload type field, keep changing.  Even if the
other fields remain constant, a packet stream with a constantly changing
SSRC field must be entered in the negative cache to avoid consuming all
of the available session contexts.  When RTP compression fails, the IP
and UDP headers may still be compressed.

In order to communicate packets in the various uncompressed and
compressed forms, this protocol depends upon the link layer being able
to provide an indication of four packet types in addition to the packet
types that indicate IPv4 and IPv6:

     FULL_HEADER - communicates the uncompressed IP header plus any fol-
     lowing headers and data to establish the uncompressed header state
     in the decompressor for a particular context.  That context is
     identified by an 8-bit session context ID.  In order to carry the
     context ID without expanding the size of the header, the ID
     replaces the low byte of the total length field in the IPv4 header
     or IPv6 base header.  (The actual length may be inferred from the
     length provided by the link layer.)  The FULL_HEADER packet type is
     part of the compression framework defined in [3], which describes
     compression of protocols other than UDP/RTP and encapsulation by
     multiple IP headers as indicated by the IPv4 protocol field or the
     IPv6 next header field.  A generation number is carried in the
     FULL_HEADER for the COMPRESSED_NON_TCP packet type defined in [3].
     The 4-bit sequence number defined in Section 3.3 of this document
     is carried in the Data field of the FULL_HEADER as defined in Sec-
     tion 5.3.2 and 12 of [3].

     COMPRESSED_NON_TCP - communicates the compressed IP and UDP headers
     as defined in [3] without compressing the IPv4 ID field.  This
     takes one or two bytes more than the COMPRESSED_UDP form listed
     next, but may be more resilient to packet loss.  This packet type
     can also carry in its Data field the 4-bit sequence number defined
     in Section 3.3.

     COMPRESSED_UDP - communicates the IP and UDP headers compressed to
     6 or fewer bytes (often 2 if UDP checksums are disabled), followed
     by any subsequent headers (possibly RTP) in uncompressed form, plus
     data.  This packet type is used when there are differences in the
     usually constant fields of the (potential) RTP header.  It rede-
     fines the SSRC field of the session context.  The format is shown
     in section 3.4.




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     COMPRESSED_RTP - indicates that the RTP header is compressed along
     with the IP and UDP headers.  The result may still be just two
     bytes.  This packet type is used when the second-order difference
     is zero and also to communicate first-order differences as delta
     encodings.  The format is shown in section 3.3.

     CONTEXT_STATE - indicates a special packet sent from the decompres-
     sor to the compressor to communicate a list of context IDs for
     which synchronization has or may have been lost.  This packet is
     only sent across the single link so it requires no IP header.  The
     format is shown in section 3.6.

Assignment of numeric codes for these four packet types in the Point-
to-Point Protocol [4] will be made by the Internet Assigned Numbers
Authority.  Section 13 of [3] specifies that the FULL_HEADER packet will
share the IPv6 packet type and demultiplex based on the IP version
field.

3.2.  Header Compression for RTP Data Packets

In the IPv4 header, only the total length, packet ID, and header check-
sum fields will normally change.  The total length is redundant with the
length provided by the link layer, and since this compression scheme
must depend upon the link layer to provide good error detection (e.g.,
PPP's CRC), the header checksum may also be elided.  This leaves only
the packet ID, which, assuming no IP fragmentation, would not need to be
communicated.  However, in order to maintain lossless compression,
changes in the packet ID will be transmitted.  In the IPv6 base header,
there is no packet ID nor header checksum and only the payload length
field changes.

In the UDP header, the length field is redundant with the IP total
length field and the length indicated by the link layer.  The UDP check-
sum field will be a constant zero if the source elects not to generate
UDP checksums.  Otherwise, the checksum must be communicated intact in
order to preserve the lossless compression.  Maintaining end-to-end
error detection for applications that require it is an important princi-
ple.

In typical usage of the RTP header, the sequence number and the times-
tamp will change from packet to packet.  If packets are not lost or
misordered, the sequence number will increment by one for each packet.
For audio packets of constant duration, the timestamp will increment by
the number of sample periods conveyed in each packet.  For video, the
timestamp will change on the first packet of each frame, but then stay
constant for any additional packets in the frame.  If each video frame
occupies only one packet, but the video frames are generated at a con-
stant rate, then again the change in the timestamp from frame to frame



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is constant.  Note that in each of these cases the second-order differ-
ence of the sequence number and timestamp fields is often zero.  When
that's not true, the magnitude of the change is usually much smaller
than the full number of bits in the field, so the size can be reduced by
encoding the difference rather than the absolute value.

The M bit will be set on the first packet of a talkspurt and the last
packet of a video frame.  If it were treated as a constant field such
that each change required sending the full RTP header, this would reduce
the compression significantly.  Therefore, one bit in the compressed
header will carry the M bit explicitly.

If the packets are flowing through an RTP mixer, most commonly for
audio, then the CSRC list and CC count will also change.  However, the
CSRC list will typically remain constant during a talkspurt or longer,
so it need be sent only when it changes.

3.3.  The protocol

When the second-order difference of the RTP header is zero, all that
need be communicated is a small sequence number to maintain synchroniza-
tion and detect packet loss between the compressor and decompressor.
Each context has its own separate sequence number space so that a single
packet loss need only invalidate one context.  To synchronize with the
decompressor, the compressor inserts the current value of the sequence
number into the Data field of the FULL_HEADER or COMPRESSED_NON_TCP
packet whenever one of those is transmitted (see Sections 5.3.2 and 6 of
[3]).

When the second-order difference of the RTP header is not zero for some
fields, the new first-order difference for just those fields is communi-
cated using a compact encoding.  The new first-order difference updates
the uncompressed header in the decompressor's session context, and it is
also stored explicitly in the context to be used for updating the field
again on subsequent packets in which the second-order difference is
zero.

In practice, the only fields for which it is useful to store the first-
order difference are the IPv4 ID field and the RTP timestamp.  For the
RTP sequence number field, the usual increment is 1.  If the sequence
number changes by other than 1, the difference must be communicated but
does not set the expected difference for the next packet.  Instead, the
expected first-order difference remains fixed at 1 so that the differ-
ence need not be explictly communicated on the next packet assuming it
is in order,

For the RTP timestamp, when a FULL_HEADER, COMPRESSED_NON_TCP or
COMPRESSED_UDP packet is sent to refresh the state, the stored first-



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order difference is initialized to zero.  If the timestamp is the same
on the next packet (e.g., same video frame), then the second-order
difference is zero.  Otherwise, the difference between the timestamps of
the two packets is transmitted as the new first-order difference.

Similarly, since the IPv4 ID field frequently increments by one, the
first-order difference for that field is initialized to one when the
state is refreshed.  Thereafter, whenever the first-order difference
changes, it is transmitted and stored in the context.

A bit mask will be used to indicate which fields have changed by other
than the expected difference.  In addition to the small link sequence
number, the list of items to be conditionally communicated in the
compressed IP/UDP/RTP header is as follows:

  I = IPv4 packet ID (always 0 if no IPv4 header)
  U = UDP checksum
  M = RTP marker bit
  S = RTP sequence number
  T = RTP timestamp
  L = RTP CSRC count and list

If 4 bits are needed for the link sequence number to get a reasonable
probability of loss detection, there are too few bits remaining to
assign one bit to each of these items and still fit them all into a sin-
gle byte to go along with the context ID.

It is not necessary to explicitly indicate the presence of the UDP
checksum because a source will typically include checksums on all pack-
ets of a session or none of them.  When the session state is initialized
with an uncompressed header, if there is a nonzero checksum present, an
unencoded 16-bit checksum will be appended to the compressed header in
all subsequent packets until this setting is changed by sending another
uncompressed packet.

Of the remaining items, the CSRC list may be the one least frequently
used.  Rather than dedicating a bit to indicate CSRC change, an unusual
combination of the other bits may be used instead.  This bit combination
is denoted MSTI.  If all four of the bits for the IP packet ID, RTP
marker bit, RTP sequence number and RTP timestamp are set, this as a
special case indicating an extended form of the compressed RTP header
will follow.  That header will include an additional byte containing the
real values of the four bits plus the CC count.  The CSRC list, of
length indicated by the CC count, will be included as in the
uncompressed header.

The following diagram shows the compressed IP/UDP/RTP header with dotted
lines indicating fields that are conditionally present.



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          0   1   2   3   4   5   6   7
        +-------------------------------+
        |        session context        |
        +---+---+---+---+---+---+---+---+
        | M | S | T | I |    sequence   |
        +---+---+---+---+---+---+---+---+
        :                               :
        +         UDP checksum          +  (implicit)
        :                               :
        +...............................+
        : M'| S'| T'| I'|      CC       :  (if MSTI)
        +...............................+
        :         delta IPv4 ID         :  (if I or I')
        +...............................+
        :      delta RTP sequence       :  (if S or S')
        +...............................+
        :      delta RTP timestamp      :  (if T or T')
        +...............................+
        :                               :
        :           CSRC list           :  (if MSTI)
        :                               :
        :                               :
        +...............................+
        :                               :
        :      RTP header extension     :  (if X set in context)
        :                               :
        :                               :
        +---+---+---+---+---+---+---+---+
        |            RTP data           |
        :                               :

The delta fields are encoded with the following variable-length mapping
for compactness:  A change of zero through 127 is represented directly
in one byte.  If the most significant two bits of the byte are 10 or 11,
this signals an extension to a two- or three-byte value, respectively.
The least significant six bits of the first byte are combined, in
decreasing order of significance, with the next one or two bytes to form
a 14- or 22- bit value.  The encoding of decimal values to hex bytes is
shown in the following table:











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      Decimal  Hex

            0  00
            :  :
          127  7F
          128  80 80
            :  :
        16383  BF FF
        16384  C0 40 00
            :  :
      4194303  FF FF FF

A change in the RTP timestamp value greater than 4194303 forces the RTP
header to be sent uncompressed using a FULL_HEADER, COMPRESSED_NON_TCP
or COMPRESSED_UDP packet type.

The context that must be maintained for each ID is as follows:

  o The full IP, UDP and RTP headers.  Multiple IP headers may be
    included on encapsulated packets.
  o The first difference for the IPv4 ID field, initialized to 1.
  o The first difference for the RTP timestamp field, initialized to 0.
  o The current 4-bit sequence number.
  o The current generation number (see [3]).


3.4.  Compression of non-RTP UDP Packets

If there is a change in any of the fields of the RTP header that are
normally constant (such as the payload type field), then an uncompressed
RTP header must be sent.  This header may be carried in a COMPRESSED_UDP
packet rather than a FULL_HEADER packet.  The COMPRESSED_UDP packet has
the same format as the COMPRESSED_RTP packet except that the M, S and T
bits are always 0 and the corresponding fields are not present:
















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          0   1   2   3   4   5   6   7
        +-------------------------------+
        |        session context        |
        +---+---+---+---+---+---+---+---+
        | 0 | 0 | 0 | I |    sequence   |
        +---+---+---+---+---+---+---+---+
        :                               :
        +         UDP checksum          +  (implicit)
        :                               :
        +...............................+
        :         delta IPv4 ID         :  (if I)
        +---+---+---+---+---+---+---+---+
        |           UDP data            |
        :   (uncompressed RTP header)   :

Note that this constitutes a form of IP/UDP header compression different
from COMPRESSED_NON_TCP packet type defined in [3].  The motivation is
to allow reaching the target of two bytes when UDP checksums are dis-
abled, as IPv4 allows.  The protocol in [3] does not use differential
coding for UDP packets, so in the IPv4 case, two bytes of IP ID, and two
bytes of UDP checksum if nonzero, would always be transmitted in addi-
tion to two bytes of compression prefix.

3.5.  Compression of RTCP Control Packets

By relying on the RTP convention that data is carried on an even port
number and the corresponding RTCP packets are carried on the next higher
(odd) port number, one could tailor separate compression schemes to be
applied to RTP and RTCP packets.  For RTCP, the compression could apply
not only to the header but also the "data", that is, the contents of the
different packet types.  The numbers in Sender Report (SR) and Receiver
Report (RR) RTCP packets would not compress well, but the text informa-
tion in the Source Description (SDES) packets could be compressed down
to a bit mask indicating each item that was present but compressed out
(for timing purposes on the SDES NOTE item and to allow the end system
to measure the average RTCP packet size for the interval calculation).

However, in the compression scheme defined here, no compression will be
done on RTCP packets for several reasons.  Since the RTP protocol
specification suggests that the RTCP packet interval be scaled so that
the aggregate RTCP bandwidth used by all participants in a session will
be no more than 5% of the session bandwidth, there is not much to be
gained from RTCP compression.  Compressing out the SDES items would
require a significant increase in the shared state that must be stored
for each context ID.  And, in order to allow compression when SDES
information for several sources was sent through an RTP "mixer", it
would be necessary to maintain a separate RTCP session context for each



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SSRC identifier.  In a session with more than 255 participants, this
would cause perfect thrashing of the context cache even when only one
participant was sending data.

3.6.  Error Recovery

Whenever the 4-bit sequence number for a particular context increments
by other than 1, except when set by a FULL_HEADER or COMPRESSED_NON_TCP
packet, the decompressor must invalidate that context and send a
CONTEXT_STATE packet back to the compressor indicating that the context
has been invalidated.  All packets for the invalid context must be dis-
carded until a FULL_HEADER or COMPRESSED_NON_TCP packet is received for
that context.

When an error occurs on the link, the link layer will usually discard
the packet that was damaged (if any), but may provide an indication of
the error.  Some time may elapse before another packet is delivered for
the same context, and then that packet would have to be discarded by the
decompressor when it is observed to be out of sequence, resulting in at
least two packets lost.  To allow faster recovery if the link does pro-
vide an explicit error indication, the decompressor may optionally send
a CONTEXT_STATE packet listing the last valid sequence number and gen-
eration number for one or more recently active contexts.  For a given
context, if the compressor has sent no compressed packet with a higher
sequence number, no corrective action is required.  Otherwise, the
compressor may mark the context invalid so that the next packet is sent
in FULL_HEADER or COMPRESSED_NON_TCP mode.  If the generation number
does not match the current generation of the COMPRESSED_NON_TCP packet,
then the FULL_HEADER must be sent.

The format of the CONTEXT_STATE packet is shown in the following
diagram.  The first byte is a type code to allow the CONTEXT_STATE
packet type to be shared for compression of other protocols in the IPv6
scheme [3].  For this IP/UDP/RTP compression scheme, the remainder of
the CONTEXT_STATE packet is structured as a list of blocks to allow the
state for multiple contexts to be indicated, preceded by a one-byte
count of the number of blocks.














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          0   1   2   3   4   5   6   7
        +---+---+---+---+---+---+---+---+
        |      compression type = 1 *   |
        +---+---+---+---+---+---+---+---+
        |         context count         |
        +---+---+---+---+---+---+---+---+
        +---+---+---+---+---+---+---+---+
        |        session context        |
        +---+---+---+---+---+---+---+---+
        | I | 0 | 0 | 0 |    sequence   |
        +---+---+---+---+---+---+---+---+
        | 0 |         generation        |
        +---+---+---+---+---+---+---+---+
                       ...
        +---+---+---+---+---+---+---+---+
        |        session context        |
        +---+---+---+---+---+---+---+---+
        | I | 0 | 0 | 0 |    sequence   |
        +---+---+---+---+---+---+---+---+
        | 0 |         generation        |
        +---+---+---+---+---+---+---+---+

    * Other compression types to be defined in [3].

The bit labeled "I" is set to one for contexts that have been marked
invalid and require a FULL_HEADER of COMPRESSED_NON_TCP packet to be
transmitted.  If the I bit is zero, the context state is advisory.

Since the CONTEXT_STATE packet itself may be lost, retransmission of one
or more blocks is allowed.  It is expected that retransmission will be
triggered only by receipt of another packet, but if the line is near
idle, retransmission might be triggered by a relatively long timer (on
the order of 1 second).

If a CONTEXT_STATE block for a given context is retransmitted, it may
cross paths with the FULL_HEADER or COMPRESSED_NON_TCP packet intended
to refresh that context.  In that case, the compressor may choose to
ignore the error indication.

In the case where UDP checksums are being transmitted, the decompressor
could attempt to use the "twice" algorithm described in section 10.1 of
[3].  In this algorithm, the delta is applied more than once on the
assumption that the delta may have been the same on the missing
packet(s) and the one subsequently received.  For the scheme defined
here, the difference in the 4-bit sequence number tells number of times
the delta must be applied.  Note, however, that there is a nontrivial
risk of an incorrect positive indication.  It may be advisable to



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request a FULL_HEADER or COMPRESSED_NON_TCP packet even if the "twice"
algorithm succeeds.

Some errors may not be detected, for example if 16 packets are lost in a
row and the link level does not provide an error indication.  In that
case, the decompressor will generate packets that are not valid.  If UDP
checksums are being transmitted, the receiver will probably detect the
invalid packets and discard them, but the receiver does not have any
means to signal the decompressor.  Therefore, it is recommended that the
decompressor verify the UDP checksum periodically, perhaps one out of 16
packets.  If an error is detected, the decompressor would invalidate the
context and signal the compressor with a CONTEXT_STATE packet.

4.  Interaction With Segmentation

A segmentation scheme may be used in conjunction with RTP header
compression to allow small, real-time packets to interrupt large,
presumably non-real-time packets in order to reduce delay.  It is
assumed that the large packets bypass the compressor and decompressor
since the interleaving would modify the sequencing of packets at the
decompressor and cause the appearance of errors.  Header compression
should be less important for large packets since the overhead ratio is
smaller.

If some packets from an RTP session context are selected for segmenta-
tion (perhaps based on size) and some are not, there is a possibility of
re-ordering.  This would reduce the compression efficiency because the
large packets would appear as lost packets in the sequence space.  How-
ever, this should not cause more serious problems because the RTP
sequence numbers should be reconstructed correctly and will allow the
application to correct the ordering.

Link errors detected by the segmentation scheme using its own sequencing
information may be indicated to the compressor with an advisory
CONTEXT_STATE message just as for link errors detected by the link layer
itself.

The context ID byte is placed first in the COMPRESSED_RTP header so that
this byte may be shared with the segmentation layer if such sharing is
feasible and has been negotiated.  Since the context ID may have any
value, it can be set to match context information from the segmentation
layer.

5.  Negotiating Compression

The use of IP/UDP/RTP compression over a particular link is a function
of the link-layer protocol.  It is expected that such negotiation will
be defined separately for PPP [4], for example.



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6.  Acknowledgments

Several people have contributed to the design of this compression scheme
and related problems.  Scott Petrack initiated discussion of RTP header
compression in the AVT working group at Los Angeles in March, 1996.
Carsten Bormann has developed an overall achitecture for compression in
combination with traffic control across a low-speed link, and made
several specific contributions to the scheme described here.  David Oran
independently developed a note based on similar ideas, and suggested the
use of PPP Multilink protocol for segmentation.  Mikael Degermark has
contributed advice on integration of this compression scheme with the
IPv6 compression framework.

7.  References:


[1] H. Shulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
    A Transport Protocol for real-time applications," RFC 1889.

[2] V. Jacobson, "TCP/IP Compression for Low-Speed Serial Links,"
    RFC 1144.

[3] M. Degermark, B. Nordgren, and S. Pink, "Header Compression for
    IPv6," work in progress.

[4] W. Simpson, "The Point-to-Point Protocol (PPP)", RFC 1548.


8.  Security Considerations

Because encryption eliminates the redundancy that this compression
scheme tries to exploit, there is some inducement to forgo encryption in
order to achieve operation over a low-bandwidth link.  However, for
those cases where encryption of data and not headers is satisfactory,
RTP does specify an alternative encryption method indicated by different
payload types.

9.  Authors' Addresses

   Stephen L. Casner
   Precept Software, Inc.
   1072 Arastradero Road
   Palo Alto, CA 94304
   United States
   EMail: casner@precept.com

   Van Jacobson
   MS 46a-1121



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   Lawrence Berkeley National Laboratory
   Berkeley, CA 94720
   United States
   EMail: van@ee.lbl.gov















































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