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RFC2508 - Compressing IP/UDP/RTP Headers for Low-Speed Serial Links

王朝other·作者佚名  2008-05-31
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Network Working Group S. Casner

Request for Comments: 2508 Cisco Systems

Category: Standards Track V. Jacobson

Cisco Systems

February 1999

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

Status of this Memo

This document specifies an Internet standards track protocol for the

Internet community, and requests discussion and suggestions for

improvements. Please refer to the current edition of the "Internet

Official Protocol Standards" (STD 1) for the standardization state

and status of this protocol. Distribution of this memo is unlimited.

Copyright Notice

Copyright (C) The Internet Society (1999). All Rights Reserved.

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).

The key Words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",

"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this

document are to be interpreted as described in RFC2119.

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. (Some existing applications operating in this environment use

an application-specific protocol with a header of a few bytes that

has reduced functionality 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

combination of IP, UDP and RTP headers on a link-by-link basis.

Compressing 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 specified in [3] for use with both IPv6 and IPv4. That

framework 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 the protocol may also be used with reduced performance

on simplex links. This compression scheme performs best on local

links with low round-trip-time.

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 compression defined here.

It should be noted that implementation simplicity is an important

factor to consider in evaluating 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. Higher

compression gain might be achieved by communicating more complex

models for the changing header fields from the compressor to the

decompressor, but that complexity is deemed unnecessary. The next

sections discuss some of the tradeoffs 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. To gain these performance

improvements, this specification includes an explicit error

indication sent on the reverse path.

On a simplex link, it would be possible to use a periodic refresh

instead. Whenever the decompressor detected an error in a particular

packet stream, it would simply discard all packets in that stream

until an uncompressed header was received for that stream, and then

resume decompression. The penalty would be the potentially large

number of packets discarded. The periodic refresh method described

in Section 3.3 of [3] applies to IP/UDP/RTP compression on simplex

links or links with high delay as well as to other non-TCP packet

streams.

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. Segmentation of large, non-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 inappropriate 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 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 RFC1144 [2].

Readers are referred to that RFCfor 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 reduction in data

rate comes from the observation 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 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 difference was zero. In that case,

the decompressor can reconstruct the original header without any loss

of information simply by adding the first-order differences to the

saved uncompressed header as each compressed packet is received.

Just as TCP/IP header compression maintains shared state for multiple

simultaneous TCP connections, this IP/UDP/RTP compression SHOULD

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. A compressor implementation might use a hash function on

these fields to index a table of stored session contexts. The

compressed packet carries a small integer, called the session context

identifier or CID, to indicate in which session context that packet

should be interpreted. The decompressor can use the CID to index its

table of stored session contexts directly.

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 heuristic gives the wrong answer. This does

require executing the compression algorithm for all UDP packets, or

at least those with even port numbers (see section 3.4).

Most compressor implementations will need to maintain a "negative

cache" of packet streams that have failed to compress as RTP packets

for some number of attempts in order to avoid further attempts.

Failing to compress means that some 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 such fields

remain constant, a packet stream with a constantly changing SSRC

field SHOULD be entered in the negative cache to avoid consuming all

of the available session contexts. The negative cache is indexed by

the source and destination IP address and UDP port pairs but not the

RTP SSRC field since the latter may be changing. When RTP

compression fails, the IP and UDP headers MAY still be compressed.

Fragmented IP Packets that are not initial fragments and packets that

are not long enough to contain a complete UDP header MUST NOT be sent

as FULL_HEADER packets. Furthermore, packets that do not

additionally contain at least 12 bytes of UDP data MUST NOT be used

to establish RTP context. If such a packet is sent as a FULL_HEADER

packet, it MAY be followed by COMPRESSED_UDP packets but MUST NOT be

followed by COMPRESSED_RTP packets.

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 [4]), 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. (Some systems increment the ID with the

bytes swapped, which results in slightly less compression.) 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 principle.

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

upstream from the compressor, 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 constant rate, then 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 an audio 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

information 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 negotiated, 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, possibly including a CSRC

list, for the last packet sent by the compressor or

reconstructed by the decompressor.

o The first-order difference for the IPv4 ID field, initialized to

1 whenever an uncompressed IP header for this context is

received and updated each time a delta IPv4 ID field is received

in a compressed packet.

o The first-order difference for the RTP timestamp field,

initialized to 0 whenever an uncompressed packet for this

context is received and updated each time a delta RTP timestamp

field is received in a compressed packet.

o The last value of the 4-bit sequence number, which is 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 set to zero if the COMPRESSED_NON_TCP packet type,

defined below, is never used.

o A context-specific delta encoding table (see section 3.3.4) may

optionally be negotiated for each context.

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 new packet formats in addition

to the normal IPv4 and IPv6 packet formats:

FULL_HEADER - communicates the uncompressed IP header plus any

following 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 header 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 by other than the expected

amount to establish the first-order differences after an

uncompressed RTP header is sent and whenever they change. The

format is shown in section 3.3.2.

CONTEXT_STATE - indicates a special packet sent from the

decompressor 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.

Assignments of numeric codes for these packet formats in the Point-

to-Point Protocol [4] are to 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 combination 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):

For 8-bit context ID:

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

01 Generation CID First length field

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

0 seq Second length field

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

For 16-bit context ID:

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

11 Generation 0 seq First length field

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

CID Second length field

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The first bit in the first length field indicates the length of the

CID. The length of the CID MUST either be constant for all contexts

or two additional distinct packet types MUST be provided to

separately indicate COMPRESSED_UDP and COMPRESSED_RTP packet formats

with 8- and 16-bit CIDs. 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 implementations that do not use

COMPRESSED_NON_TCP packets, the compressor SHOULD set the generation

value to zero. For consistent 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 is 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 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 the session context identifier and a small sequence

number (not related to the RTP sequence number) to maintain

synchronization and detect packet loss between the compressor and

decompressor.

If the second-order difference of the RTP header is not zero for some

fields, the new first-order difference for just those fields is

communicated using a compact encoding. The new first-order

difference values are added to the corresponding fields in the

uncompressed header in the decompressor's session context, and are

also stored explicitly in the context to be added to the

corresponding fields again on each subsequent packet in which the

second-order difference is zero. Each time the first-order

difference changes, it is transmitted and stored in the context.

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 difference need not be explicitly 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 to be added to the timestamp in the uncompressed

header stored in the decompressor's context and also stored as the

first-order difference in that context. Each time the first-order

difference changes on subsequent packets, that difference is again

transmitted and used to update the context.

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 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

single byte to go along with the context ID.

It is not necessary to explicitly carry the U bit to indicate the

presence of the UDP checksum because a source will typically include

check-sums on all packets 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

inserted into the compressed header in all subsequent packets until

this setting is changed by sending another uncompressed packet.

Of the remaining items, the L bit for the CSRC count and list may be

the one least frequently used. Rather than dedicating a bit in the

mask 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 is 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 other fields of the RTP header (version, P bit, X bit, payload

type and SSRC identifier) are assumed to remain relatively constant.

In particular, the SSRC identifier is defined to be constant for a

given context because it is one of the factors selecting the context.

If any of the other fields change, the uncompressed RTP header MUST

sent as described in Section 3.3.3.

The following diagram shows the compressed IP/UDP/RTP header with

dotted lines indicating fields that are conditionally present. The

most significant bit is numbered 0. Multi-byte fields are sent in

network byte order (most significant byte first). The delta fields

are often a single byte as shown but may be two or three bytes

depending upon the delta value as explained in Section 3.3.4.

0 1 2 3 4 5 6 7

+...............................+

: msb of session context ID : (if 16-bit CID)

+-------------------------------+

lsb of session context ID

+---+---+---+---+---+---+---+---+

M S T I link sequence

+---+---+---+---+---+---+---+---+

: :

+ UDP checksum + (if nonzero in context)

: :

+...............................+

: :

+ "RANDOM" fields + (if encapsulated)

: :

+...............................+

: M' S' T' I' CC : (if MSTI = 1111)

+...............................+

: delta IPv4 ID : (if I or I' = 1)

+...............................+

: delta RTP sequence : (if S or S' = 1)

+...............................+

: delta RTP timestamp : (if T or T' = 1)

+...............................+

: :

: CSRC list : (if MSTI = 1111

: : and CC nonzero)

: :

+...............................+

: :

: RTP header extension : (if X set in context)

: :

: :

+-------------------------------+

RTP data

/ /

/ /

+-------------------------------+

: padding : (if P set in context)

+...............................+

When more than one IPv4 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 UDP

checksum if present or the MSTI byte if not, as shown in the diagram.

Only if an IPv4 packet immediately precedes 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 preceding the UDP

header, then the I bit MUST be 0 and no delta IPv4 ID field will be

present.

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

+...............................+

: msb of session context ID : (if 16-bit CID)

+-------------------------------+

lsb of session context ID

+---+---+---+---+---+---+---+---+

0 0 0 I link sequence

+---+---+---+---+---+---+---+---+

: :

+ UDP checksum + (if nonzero in context)

: :

+...............................+

: :

+ "RANDOM" fields + (if encapsulated)

: :

+...............................+

: delta IPv4 ID : (if I = 1)

+-------------------------------+

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 disabled, 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 addition 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 this

default table and Huffman encoding are examples, allow a reasonable

table size 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 the order in which they

are transmitted (network byte order). The first and last values in

each contiguous range are shown, with ellipses 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 combined, 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

intentionally out-of-order RTP timestamps on MPEG video [5]. These

events are less likely, so a smaller range of negative values is

encoded using otherwise 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).

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 discarded until a FULL_HEADER or

COMPRESSED_NON_TCP packet is received for that context to re-

establish consistent state (unless the "twice" algorithm is used as

described later in this section). 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 provide an explicit error indication, the

decompressor MAY optionally send an advisory CONTEXT_STATE packet

listing the last valid sequence number and generation number for one

or more recently active contexts (not necessarily all). For a given

context, if the compressor has sent no compressed packet with a

higher sequence number, and if the generation number matches the

current generation, no corrective action is required. Otherwise, the

compressor MAY choose to mark the context invalid so that the next

packet is sent in FULL_HEADER or COMPRESSED_NON_TCP mode (FULL_HEADER

is required if the generation doesn't match). However, note that if

the link round-trip-time is large compared to the inter-packet

spacing, there may be several packets from multiple contexts in

flight across the link, increasing the probability that the sequence

numbers will already have advanced when the CONTEXT_STATE packet is

received by the compressor. The result could be that some contexts

are invalidated unnecessarily, causing extra bandwidth to be

consumed.

The format of the CONTEXT_STATE packet is shown in the following

diagrams. The first byte is a type code to allow the CONTEXT_STATE

packet type to be shared by multiple compression schemes within the

general compression framework specified in [3]. The contents of the

remainder of the packet depends upon the compression scheme. For the

IP/UDP/RTP compression scheme specified here, 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.

Two type code values are used for the IP/UDP/RTP compression scheme.

The value 1 indicates that 8-bit session context IDs are being used:

0 1 2 3 4 5 6 7

+---+---+---+---+---+---+---+---+

1 = IP/UDP/RTP with 8-bit CID

+---+---+---+---+---+---+---+---+

context count

+---+---+---+---+---+---+---+---+

+---+---+---+---+---+---+---+---+

session context ID

+---+---+---+---+---+---+---+---+

I 0 0 0 sequence

+---+---+---+---+---+---+---+---+

0 0 generation

+---+---+---+---+---+---+---+---+

...

+---+---+---+---+---+---+---+---+

session context ID

+---+---+---+---+---+---+---+---+

I 0 0 0 sequence

+---+---+---+---+---+---+---+---+

0 0 generation

+---+---+---+---+---+---+---+---+

The value 2 indicates that 16-bit session context IDs are being used.

The session context ID is sent in network byte order (most

significant byte first):

0 1 2 3 4 5 6 7

+---+---+---+---+---+---+---+---+

2 = IP/UDP/RTP with 16-bit CID

+---+---+---+---+---+---+---+---+

context count

+---+---+---+---+---+---+---+---+

+---+---+---+---+---+---+---+---+

+ session context ID +

+---+---+---+---+---+---+---+---+

I 0 0 0 sequence

+---+---+---+---+---+---+---+---+

0 0 generation

+---+---+---+---+---+---+---+---+

...

+---+---+---+---+---+---+---+---+

+ session context ID +

+---+---+---+---+---+---+---+---+

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.

The I bit is set to zero to indicate advisory context state that MAY

be sent following a link error indication.

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 MAY 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 MAY 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. Success is

indicated by a checksum match. 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.

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 information 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 the RTCP headers and "data" for several reasons (though

compression SHOULD still be applied to the IP and UDP headers).

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.

Even though RTCP is not compressed, the fraction of the total

bandwidth occupied by RTCP packets on the compressed link remains no

more than 5% in most cases, assuming that the RTCP packets are sent

as COMPRESSED_UDP packets. Given that the uncompressed RTCP traffic

consumes no more than 5% of the total session bandwidth, then for a

typical RTCP packet length of 90 bytes, the portion of the compressed

bandwidth used by RTCP will be no more than 5% if the size of the

payload in RTP data packets is at least 108 bytes. If the size of

the RTP data payload is smaller, the fraction will increase, but is

still less than 7% for a payload size of 37 bytes. For large data

payloads, the compressed RTCP fraction is less than the uncompressed

RTCP fraction (for example, 4% at 1000 bytes).

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, 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

segmentation (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. However, 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 compressor

may assign context ID values arbitrarily, the value can be set to

match the context identifier 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 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 architecture

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] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:

A Transport Protocol for real-time applications", RFC1889,

January 1996.

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

RFC1144, February 1990.

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

IPv6", RFC2507, February 1999.

[4] Simpson, W., "The Point-to-Point Protocol (PPP)", STD 51, RFC

1661, July 1994.

[5] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP

Payload Format for MPEG1/MPEG2 Video", RFC2250, January 1998.

8. Security Considerations

Because encryption eliminates the redundancy that this compression

scheme tries to exploit, there is some inducement to forego

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.

A malfunctioning or malicious compressor could cause the decompressor

to reconstitute packets that do not match the original packets but

still have valid IP, UDP and RTP headers and possibly even valid UDP

check-sums. Such corruption may be detected with end-to-end

authentication and integrity mechanisms which will not be affected by

the compression. Constant portions of authentication headers will be

compressed as described in [3].

No authentication is performed on the CONTEXT_STATE control packet

sent by this protocol. An attacker with Access to the link between

the decompressor and compressor could inject false CONTEXT_STATE

packets and cause compression efficiency to be reduced, probably

resulting in congestion on the link. However, an attacker with

access to the link could also disrupt the traffic in many other ways.

A potential denial-of-service threat exists when using compression

techniques that have non-uniform receiver-end computational load.

The attacker can inject pathological datagrams into the stream which

are complex to decompress and cause the receiver to be overloaded and

degrading processing of other streams. However, this compression

does not exhibit any significant non-uniformity.

A security review of this protocol found no additional security

considerations.

9. Authors' Addresses

Stephen L. Casner

Cisco Systems, Inc.

170 West Tasman Drive

San Jose, CA 95134-1706

United States

EMail: casner@cisco.com

Van Jacobson

Cisco Systems, Inc.

170 West Tasman Drive

San Jose, CA 95134-1706

United States

EMail: van@cisco.com

10. Full Copyright Statement

Copyright (C) The Internet Society (1999). All Rights Reserved.

This document and translations of it may be copied and furnished to

others, and derivative works that comment on or otherwise explain it

or assist in its implementation may be prepared, copied, published

and distributed, in whole or in part, without restriction of any

kind, provided that the above copyright notice and this paragraph are

included on all such copies and derivative works. However, this

document itself may not be modified in any way, such as by removing

the copyright notice or references to the Internet Society or other

Internet organizations, except as needed for the purpose of

developing Internet standards in which case the procedures for

copyrights defined in the Internet Standards process must be

followed, or as required to translate it into languages other than

English.

The limited permissions granted above are perpetual and will not be

revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an

"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING

TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING

BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION

HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF

MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

 
 
 
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