Internet Engineering Task Force Audio/Video Transport Working Group
INTERNET-DRAFT S. Casner / Precept Software
draft-ietf-avt-crtp-02.txt V. Jacobson / LBNL
March 26, 1997
Expires: 9/97
Compressing IP/UDP/RTP Headers for Low-Speed Serial Links
Status of this Memo
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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).
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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. The IP/UDP/RTP compression defined here is
intended to fit within the more general compression framework [3] speci-
fied by Mikael Degermark, et. al., for both IPv6 and IPv4. That frame-
work defines TCP and non-TCP as two classes of transport above IP. This
specification creates IP/UDP/RTP as a third class extracted 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 would be inap-
propriate to integrate segmentation and compression in such a way that
the compression could not be used by itself in situations where segmen-
tation 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 IP and TCP headers remain constant
over the life of the connection. After sending the uncompressed header
once, these fields may be elided from the compressed headers that fol-
low. 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 proto-
col.
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. In that case, the decompressor can reconstruct the ori-
ginal header without any loss of information simply by adding the first
order differences to the saved uncompressed header.
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. The compressed
packet carries a small integer to indicate in which session context that
packet should be interpreted.
Because the RTP compression is lossless, it may be applied to any UDP
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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 execut-
ing the compression algorithm for all UDP packets. Most implementations
will need to maintain a negative cache of packet streams (identified by
IP address and UDP port pairs but not the RTP SSRC field) that have
failed to compress as RTP packets for some number of attempts. Failing
to compress means that the fields in the potential RTP header 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 con-
texts. When RTP compression fails, the IP and UDP headers may still be
compressed.
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. The packet ID usually
increments by one or a small number for each packet. 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 the RTP header, the SSRC identifier is constant in a given context
since that is part of what identifies the particular context. For most
packets, only the sequence number and the timestamp 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 con-
stant 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 addi-
tional packets in the frame. If each video frame occupies only one
packet, but the video frames are generated at a constant rate, then
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again the change in the timestamp from frame to frame is constant. Note
that in each of these cases the second-order difference of the sequence
number and timestamp fields is zero, so the next packet header can be
constructed from the previous packet header by adding the first-order
differences for these fields that are stored in the session context
along with the previous uncompressed header. When the second-order
difference is not zero, 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 new first-order difference and transmitting it
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
The compression protocol must maintain a collection of shared informa-
tion in a consistent state between the compressor and decompressor.
There is a separate session context for each IP/UDP/RTP packet stream,
as defined by a particular combination of the IP source and destination
addresses, UDP source and destination ports, and the RTP SSRC field.
The number of session contexts to be maintained may be negotiated
between the compressor and decompressor. Each context is identified by
an 8- or 16-bit identifier, depending upon the number of contexts nego-
tiated, so the maximum number is 65536. Both uncompressed and
compressed packets must carry the context ID and a 4-bit sequence number
used to 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.
The shared information in each context consists of the following items:
o The full IP, UDP and RTP headers for the last packet sent by the
compressor or reconstructed by the decompressor.
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 last value of the 4-bit sequence number used to detect packet
loss between the compressor and decompressor.
o The current generation number for non-differential coding of UDP
packets with IPv6(see [3]). For IPv4, the generation number may be
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set to zero.
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 normal IPv4 and IPv6 packets:
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. The FULL-HEADER
packet also carries the 8- or 16-bit session context identifier and
the 4-bit sequence number to establish synchronization between the
compressor and decompressor. The format is shown in section 3.3.1.
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. The RTP
header includes a potentially changed value of the SSRC field, so
this packet may redefine the session context. The format is shown
in section 3.3.3.
COMPRESSED_RTP - indicates that the RTP header is compressed along
with the IP and UDP headers. The size of this packet may still be
just two bytes, or more if differences must be communicated. This
packet type is used when the second-order difference (at least in
the usually constant fields) is zero. It includes delta encodings
for those fields that have changed, and also establishes the
first-order differences after an uncompressed RTP header is sent.
The format is shown in section 3.3.2.
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 point-to-point link so it requires no IP
header. The format is shown in section 3.3.5.
When this compression scheme is used with IPv6 as part of the general
header compression framework specified in [3], another packet type may
be used:
COMPRESSED_NON_TCP - communicates the compressed IP and UDP headers
as defined in [3] without differential encoding. If it were used
for IPv4, it would require one or two bytes more than the
COMPRESSED_UDP form listed above in order to carry the IPv4 ID
field. For IPv6, there is no ID field and this non-differential
compression is more resilient to packet loss.
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Assignment of numeric codes for these packet types in the Point-to-Point
Protocol [4] will be made by the Internet Assigned Numbers Authority.
3.3.1. FULL_HEADER (uncompressed) packet format
The definition of the FULL_HEADER packet given here is intended to be
the consistent with the definition given in [3]. Full details on design
choices are given there.
The format of the FULL_HEADER packet is the same as that of the original
packet. In the IPv4 case, this is usually an IP header, followed by a
UDP header and UDP payload that may be an RTP header and its payload.
However, the FULL_HEADER packet may also carry IP encapsulated packets,
in which case there would be two IP headers followed by UDP and possibly
RTP. Or in the case of IPv6, the packet may be built of some combina-
tion of IPv6 and IPv4 headers. Each successive header is indicated by
the type field of the previous header, as usual.
The FULL_HEADER packet differs from the corresponding normal IPv4 or
IPv6 packet in that it must also carry the compression context ID and
the 4-bit sequence number. In order to avoid expanding the size of the
header, these values are inserted into length fields in the IP and UDP
headers since the actual length may be inferred from the length provided
by the link layer. Two 16-bit length fields are needed; these are taken
from the first two available headers in the packet. That is, for an
IPv4/UDP packet, the first length field is the total length field of the
IPv4 header, and the second is the length field of the UDP header. For
an IPv4 encapsulated packet, the first length field would come from the
total length field of the first IP header, and the second length field
would come from the total length field of the second IP header.
As specified in Sections 5.3.2 of [3], the position of the context ID
(CID) and 4-bit sequence number varies depending upon whether 8- or 16-
bit context IDs have been selected, as shown in the following diagram
(16 bits wide, with the most-significant bit is to the left):
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For 8-bit context ID:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0|1| Generation| CID | First length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 0 | seq | Second length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
For 16-bit context ID:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|1| Generation| 0 | seq | First length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CID | Second length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The first bit int he first length field indicates the length of the CID.
The second bit in the first length field is 1 to indicate that the 4-bit
sequence number is present, as is always the case for this IP/UDP/RTP
compression scheme. The generation field is used with IPv6 for
COMPRESSED_NON_TCP packets as described in [3]. For IPv4-only implemen-
tations, the compressor may set the generation value to zero. For con-
sistent operation between IPv4 and IPv6, the generation value is stored
in the context when it is received by the decompressor, and the most
recent value is returned in the CONTEXT_STATE packet.
When a FULL_HEADER packet is received, the complete set of headers is
stored into the context selected by the context ID. The 4-bit sequence
number is also stored in the context, thereby resynchronizing the
decompressor to the compressor.
When COMPRESSED_NON_TCP packets are used, the 4-bit sequence number is
inserted into the "Data Field" of that packet and the D bit is set as
described in Section 6 of [3]. When a COMPRESSED_NON_TCP packet is
received, the generation number must be compared to the value stored in
the context. If they are not the same, the context is not up to date
and must be refreshed by a FULL_HEADER packet. If the generation does
match, then the compressed IP and UDP header information, the 4-bit
sequence number, and the (potential) RTP header are all stored into the
saved context.
The amount of memory required to store the context will vary depending
upon how many encapsulating headers are included in the FULL_HEADER
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packet. The compressor and decompressor may negotiate a maximum header
size.
3.3.2. COMPRESSED_RTP packet format
When the second-order difference of the RTP header from packet to packet
is zero, the decompressor can reconstruct a packet simply by adding the
stored first-order differences to the stored uncompressed header
representing the previous packet. All that need be communicated is a
small sequence number to maintain synchronization and detect packet loss
between the compressor and decompressor.
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 values
are used to update the uncompressed header in the decompressor's session
context, and are also stored explicitly in the context to be used for
updating the fields 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 RTP state, the stored
first-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 by a FULL_HEADER packet, or when a COMPRESSED_NON_TCP
packet is sent since it carries the ID field in uncompressed form.
Thereafter, whenever the first-order difference changes, it is transmit-
ted 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:
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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 just as it appears in
the uncompressed RTP header.
The following diagram shows the compressed IP/UDP/RTP header with dotted
lines indicating fields that are conditionally present. The most signi-
ficant bit is numbered 0.
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0 1 2 3 4 5 6 7
+-------------------------------+
| session context |
+---+---+---+---+---+---+---+---+
| M | S | T | I | sequence |
+---+---+---+---+---+---+---+---+
: :
+ "RANDOM" fields + (only if encapsulated)
: :
+...............................+
: :
+ 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 |
: :
When more than one IP header is present in the context as initialized
by the FULL_HEADER packet, then the IP ID fields of encapsulating
headers must be sent as absolute values as described in [3]. These
fields are identified as "RANDOM" fields. They are inserted into the
COMPRESSED_RTP packet in the same order as they appear in the original
headers, immediately following the MSTI byte as shown. Only if an
IPv4 packet immediately preceeds the UDP header will the IP ID of that
header be sent differentially, i.e., potentially with no bits if the
second difference is zero, or as a delta IPv4 ID field if not. If
there is not an IPv4 header immediately preceeding the UDP header,
then the I bit will be 0 and no delta IPv4 ID field will be present.
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3.3.3. COMPRESSED_UDP packet format
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. If the IP and UDP headers do not also require
updating, this RTP 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 delta fields are never included:
0 1 2 3 4 5 6 7
+-------------------------------+
| session context |
+---+---+---+---+---+---+---+---+
| 0 | 0 | 0 | I | sequence |
+---+---+---+---+---+---+---+---+
: :
+ "RANDOM" fields + (only if encapsulated)
: :
+...............................+
: :
+ 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. For IPv6, the
COMPRESSED_NON_TCP packet type may be used instead.
3.3.4. Encoding of differences
The delta fields in the COMPRESSED_RTP and COMPRESSED_UDP packets are
encoded with a variable-length mapping for compactness of the more
commonly-used values. A default encoding is specified below, but it is
recommended that implementations use a table-driven delta encoder and
decoder to allow negotiation of a table specific for each session if
appropriate, possibly even an optimal Huffman encoding. Encodings based
on sequential interpretation of the bit stream, of which the default
table and Huffman encoding are examples, allow a reasonable table size
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and may result in an execution speed faster than a non-table-driven
implementation with explicit tests for ranges of values.
The default delta encoding is specified in the following table. This
encoding was designed to efficiently encode the small changes that may
occur in the IP ID and in RTP sequence number when packets are lost
upstream from the compressor, yet still handling most audio and video
deltas in two bytes. The column on the left is the decimal value to be
encoded, and the column on the right is the resulting sequence of bytes
shown in hexadecimal and in network byte order. The first and last
values in each contiguous range are shown, with elipses in between:
Decimal Hex
-16384 C0 00 00
: :
-129 C0 3F 7F
-128 80 00
: :
-1 80 7F
0 00
: :
127 7F
128 80 80
: :
16383 BF FF
16384 C0 40 00
: :
4194303 FF FF FF
For positive values, 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 com-
bined, in decreasing order of significance, with the next one or two
bytes to form a 14- or 22- bit value.
Negative deltas may occur when packets are misordered or in the inten-
tionally out-of-order RTP timestamps on MPEG video. These events are
less likely, so a smaller range of negative values is encoded using oth-
erwise redundant portions of the positive part of the table.
A change in the RTP timestamp value less than -16384 or greater than
4194303 forces the RTP header to be sent uncompressed using a
FULL_HEADER, COMPRESSED_NON_TCP or COMPRESSED_UDP packet type. The IP
ID and RTP sequence number fields are only 16 bits, so negative deltas
for those fields should be masked to 16 bits and then encoded (as large
positive 16-bit numbers).
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3.3.5. 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 to re-establish consistent state. Since multiple
compressed packets may arrive in the interim, the decompressor should
not retransmit the CONTEXT_STATE packet for every compressed packet
received, but instead should limit the rate of retransmission to avoid
flooding the reverse channel.
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 schemes for other protocols
that may be defined in parallel with this one. For this IP/UDP/RTP
compression scheme that type code has the value 1, and 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
+---+---+---+---+---+---+---+---+
| IP/UDP/RTP compression = 1 |
+---+---+---+---+---+---+---+---+
| context count |
+---+---+---+---+---+---+---+---+
+---+---+---+---+---+---+---+---+
| session context |
+---+---+---+---+---+---+---+---+
| I | 0 | 0 | 0 | sequence |
+---+---+---+---+---+---+---+---+
| 0 | 0 | generation |
+---+---+---+---+---+---+---+---+
...
+---+---+---+---+---+---+---+---+
| session context |
+---+---+---+---+---+---+---+---+
| I | 0 | 0 | 0 | sequence |
+---+---+---+---+---+---+---+---+
| 0 | 0 | generation |
+---+---+---+---+---+---+---+---+
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
request a FULL_HEADER or COMPRESSED_NON_TCP packet even if the "twice"
algorithm succeeds.
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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.
3.4. 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
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.
3.5. Compression of non-RTP UDP Packets
As described earlier, the COMPRESSED_UDP packet may be used to compress
UDP packets that don't carry RTP. Whatever data follows the UDP header
is unlikely to have some constant values in the bits that correspond to
usually constant fields in the RTP header. In particular, the SSRC
field would likely change. Therefore, it is necessary to keep track of
the non-RTP UDP packet streams to avoid using up all the context slots
as the "SSRC field" changes (since that field is part of what identifies
a particular RTP context). Those streams may each be given a context,
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but the encoder would set a flag in the context to indicate that the
changing SSRC field should be ignored and COMPRESSED_UDP packets should
always be sent instead of COMPRESSED_RTP packets.
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. The following items may
be negotiated:
o The size of the context ID.
o The maximum size of the stack of headers in the context.
o A context-specific table for decoding of delta values.
6. Acknowledgments
Several people have contributed to the design of this compression scheme
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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. Schulzrinne, 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 in which only the RTP
payload is encrypted and the headers are left in the clear. That would
allow compression to still be applied.
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
Lawrence Berkeley National Laboratory
Berkeley, CA 94720
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United States
EMail: van@ee.lbl.gov
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