RFC2689 - Providing Integrated Services over Low-bitrate Links

Network Working Group C. Bormann

Request for Comments: 2689 Universitaet Bremen TZI

Category: Informational September 1999

Providing Integrated Services over Low-bitrate Links

Status of this Memo

This memo provides information for the Internet community. It does

not specify an Internet standard of any kind. Distribution of this

memo is unlimited.

Abstract

This document describes an architecture for providing integrated

services over low-bitrate links, sUCh as modem lines, ISDN B-

channels, and sub-T1 links. It covers only the lower parts of the

Internet Multimedia Conferencing Architecture [1]; additional

components required for application services such as Internet

Telephony (e.g., a session initiation protocol) are outside the scope

of this document. The main components of the architecture are: a

real-time encapsulation format for asynchronous and synchronous low-

bitrate links, a header compression architecture optimized for real-

time flows, elements of negotiation protocols used between routers

(or between hosts and routers), and announcement protocols used by

applications to allow this negotiation to take place.

1. Introduction

As an extension to the "best-effort" services the Internet is well-

known for, additional types of services ("integrated services") that

support the transport of real-time multimedia information are being

developed for, and deployed in the Internet. Important elements of

this development are:

- parameters for forwarding mechanisms that are appropriate for

real-time information [11, 12],

- a setup protocol that allows establishing special forwarding

treatment for real-time information flows (RSVP [4]),

- a transport protocol for real-time information (RTP/RTCP [6]).

In addition to these elements at the network and transport levels of

the Internet Multimedia Conferencing Architecture [1], further

components are required to define application services such as

Internet Telephony, e.g., protocols for session initiation and

control. These components are outside the scope of this document.

Up to now, the newly developed services could not (or only very

inefficiently) be used over forwarding paths that include low-bitrate

links such as 14.4, 33.6, and 56 kbit/s modems, 56 and 64 kbit/s ISDN

B-channels, or even sub-T1 links. The encapsulation formats used on

these links are not appropriate for the simultaneous transport of

arbitrary data and real-time information that has to meet stringent

delay requirements. Transmission of a 1500 byte packet on a 28.8

kbit/s modem link makes this link unavailable for the transmission of

real-time information for about 400 ms. This adds a worst-case delay

that causes real-time applications to operate with round-trip delays

on the order of at least a second -- unacceptable for real-time

conversation. In addition, the header overhead associated with the

protocol stacks used is prohibitive on low-bitrate links, where

compression down to a few dozen bytes per real-time information

packet is often desirable. E.g., the overhead of at least 44

(4+20+8+12) bytes for HDLC/PPP, IP, UDP, and RTP completely

overshadows typical audio payloads such as the 19.75 bytes needed for

a G.723.1 ACELP audio frame -- a 14.4 kbit/s link is completely

consumed by this header overhead alone at 40 real-time frames per

second total (i.e., at 25 ms packetization delay for one stream or 50

ms for two streams, with no space left for data, yet). While the

header overhead can be reduced by combining several real-time

information frames into one packet, this increases the delay incurred

while filling that packet and further detracts from the goal of

real-time transfer of multi-media information over the Internet.

This document describes an approach for addressing these problems.

The main components of the architecture are:

- a real-time encapsulation format for asynchronous and synchronous

low-bitrate links,

- a header compression architecture optimized for real-time flows,

- elements of negotiation protocols used between routers (or between

hosts and routers), and

- announcement protocols used by applications to allow this

negotiation to take place.

2. Design Considerations

The main design goal for an architecture that addresses real-time

multimedia flows over low-bitrate links is that of minimizing the

end-to-end delay. More specifically, the worst case delay (after

removing possible outliers, which are equivalent to packet losses

from an application point of view) is what determines the playout

points selected by the applications and thus the delay actually

perceived by the user.

In addition, any such architecture should obviously undertake every

attempt to maximize the bandwidth actually available to media data;

overheads must be minimized.

An important component of the integrated services architecture is the

provision of reservations for real-time flows. One of the problems

that systems on low-bitrate links (routers or hosts) face when

performing admission control for such reservations is that they must

translate the bandwidth requested in the reservation to the one

actually consumed on the link. Methods such as data compression

and/or header compression can reduce the requirements on the link,

but admission control can only make use of the reduced requirements

in its calculations if it has enough information about the data

stream to know how effective the compression will be. One goal of

the architecture therefore is to provide the integrated services

admission control with this information. A beneficial side effect

may be to allow the systems to perform better compression than would

be possible without this information. This may make it worthwhile to

provide this information even when it is not intended to make a

reservation for a real-time flow.

3. The Need for a Concerted Approach

Many technical approaches come to mind for addressing these problems,

in particular a new form of low-delay encapsulation to address delay

and header compression methods to address overhead. This section

shows that these techniques should be combined to solve the problem.

3.1. Real-Time Encapsulation

The purpose of defining a real-time link-layer encapsulation protocol

is to be able to introduce newly arrived real-time packets into the

link-layer data stream without having to wait for the currently

transmitted (possibly large) packet to end. Obviously, a real-time

encapsulation must be part of any complete solution as the problem of

delays induced by large frames on the link can only be solved on this

layer.

To be able to switch to a real-time packet quickly in an interface

driver, it is first necessary to identify packets that belong to

real-time flows. This can be done using a heuristic approach (e.g.,

favor the transmission of highly periodic flows of small packets

transported in IP/UDP, or use the IP precedence fields in a specific

way defined within an organization). Preferably, one also could make

use of a protocol defined for identifying flows that require special

treatment, i.e. RSVP. Of the two service types defined for use with

RSVP now, the guaranteed service will only be available in certain

environments; for this and various other reasons, the service type

chosen for many adaptive audio/video applications will most likely be

the controlled-load service. Controlled-load does not provide

control parameters for target delay; thus it does not unambiguously

identify those packet streams that would benefit most from being

transported in a real-time encapsulation format. This calls for a

way to provide additional parameters in integrated services flow

setup protocols to control the real-time encapsulation.

Real-time encapsulation is not sufficient on its own, however: Even

if the relevant flows can be appropriately identified for real-time

treatment, most applications simply cannot operate properly on low-

bitrate links with the header overhead implied by the combination of

HDLC/PPP, IP, UDP, and RTP, i.e. they absolutely require header

compression.

3.2. Header Compression

Header compression can be performed in a variety of elements and at a

variety of levels in the protocol architecture. As many vendors of

Internet Telephony products for PCs ship applications, the approach

that is most obvious to them is to reduce overhead by performing

header compression at the application level, i.e. above transport

protocols such as UDP (or actually by using a non-standard,

efficiently coded header in the first place).

Generally, header compression operates by installing state at both

ends of a path that allows the receiving end to reconstruct

information omitted at the sending end. Many good techniques for

header compression (RFC1144, [2]) operate on the assumption that the

path will not reorder the frames generated. This assumption does not

hold for end-to-end compression; therefore additional overhead is

required for resequencing state changes and for compressed packets

making use of these state changes.

Assume that a very good application level header compression solution

for RTP flows could be able to save 11 out of the 12 bytes of an RTP

header [3]. Even this perfect solution only reduces the total header

overhead by 1/4. It would have to be deployed in all applications,

even those that operate on systems that are attached to higher-

bitrate links.

Because of this limited effectiveness, the AVT group that is

responsible for RTP within the IETF has decided to not further pursue

application level header compression.

For router and IP stack vendors, the obvious approach is to define

header compression that can be negotiated between peer routers.

Advanced header compression techniques now being defined in the IETF

[2] certainly can relieve the link from significant parts of the

IP/UDP overhead (i.e., most of 28 of the 44 bytes mentioned above).

One of the design principles of the new IP header compression

developed in conjunction with IPv6 is that it stops at layers the

semantics of which cannot be inferred from information in lower layer

(outer) headers. Therefore, this header compression technique alone

cannot compress the data that is contained within UDP packets.

Any additional header compression technique runs into a problem: If

it assumes specific application semantics (i.e., those of RTP and a

payload data format) based on heuristics, it runs the risk of being

triggered falsely and (e.g. in case of packet loss) reconstructing

packets that are catastrophically incorrect for the application

actually being used. A header compression technique that can be

operated based on heuristics but does not cause incorrect

decompression even if the heuristics failed is described in [7]; a

companion document describes the mapping of this technique to PPP

[10].

With all of these techniques, the total IP/UDP/RTP header overhead

for an audio stream can be reduced to two bytes per packet. This

technology need only be deployed at bottleneck links; high-speed

links can transfer the real-time streams without routers or switches

eXPending CPU cycles to perform header compression.

4. Principles of Real-Time Encapsulation for Low-Bitrate Links

The main design goal for a real-time encapsulation is to minimize the

delay incurred by real-time packets that become available for sending

while a long data packet is being sent. To achieve this, the

encapsulation must be able to either abort or suspend the transfer of

the long data packet. As an additional goal is to minimize the

overhead required for the transmission of packets from periodic

flows, this strongly argues for being able to suspend a packet, i.e.

segment it into parts between which the real-time packets can be

transferred.

4.1. Using existing IP fragmentation

Transmitting only part of a packet, to allow higher-priority traffic

to intervene and then resuming its transmission later on, is a kind

of fragmentation. Fragmentation is an existing functionality of the

IP layer: An IPv4 header already contains fields that allow a large

IP datagram to be fragmented into small parts. A sender's "real-time

PPP" implementation might simply indicate a small MTU to its IP stack

and thus cause all larger datagrams to be fragmented down to a size

that allows the Access delay goals to be met (this assumes that the

IP stack is able to priority-tag fragments, or that the PPP

implementation is able to correlate the fragments to the initial one

that carries the information relevant for prioritizing, or that only

initial fragments can be high-priority). (Also, a PPP implementation

can negotiate down the MTU of its peer, causing the peer to fragment

to a small size, which might be considered a crude form of

negotiating an access delay goal with the peer system -- if that

system supports priority queueing at the fragment level.)

Unfortunately, a full, 20 byte IP header is needed for each fragment

(larger when IP options are used). This limits the minimum size of

fragments that can be used without too much overhead. (Also, the

size of non-final fragments must be a multiple of 8 bytes, further

limiting the choice.) With path MTU discovery, IP level

fragmentation causes TCP implementations to use small MSSs -- this

further increases the per-packet overhead to 40 bytes per fragment.

In any case, fragmentation at the IP level persists on the path

further down to the datagram receiver, increasing the transmission

overheads and router load throughout the network. With its high

overhead and the adverse effect on the Internet, IP level

fragmentation can only be a stop-gap mechanism when no other

fragmentation protocol is available in the peer implementation.

4.2. Link-Layer Mechanisms

Cell-oriented multiplexing techniques such as ATM that introduce

regular points where cells from a different packet can be

interpolated are too inefficient for low-bitrate links; also, they

are not supported by chips used to support the link layer in low-

bitrate routers and host interfaces.

Instead, the real-time encapsulation should as far as possible make

use of the capabilities of the chips that have been deployed. On

synchronous lines, these chips support HDLC framing; on asynchronous

lines, an asynchronous variant of HDLC that usually is implemented in

software is being used. Both variants of HDLC provide a delimiting

mechanism to indicate the end of a frame over the link. The obvious

solution to the segmentation problem is to combine this mechanism

with an indication of whether the delimiter terminates or suspends

the current packet.

This indication could be in an octet appended to each frame

information field; however, seven out of eight bits of the octet

would be wasted. Instead, the bit could be carried at the start of

the next frame in conjunction with multiplexing information (PPP

protocol identifier etc.) that will be required here anyway. Since

the real-time flows will in general be periodic, this multiplexing

information could convey (part of) the compressed form of the header

for the packet. If packets from the real-time flow generally are of

constant length (or have a defined maximum length that is often

used), the continuation of the suspended packet could be immediately

attached to it, without expending a further frame delimiter, i.e.,

the interpolation of the real-time packet would then have zero

overhead. Since packets from low-delay real-time flows generally

will not require the ability to be further suspended, the

continuation bit could be reserved for the non-real-time packet

stream.

One real-time encapsulation format with these (and other) functions

is described in ITU-T H.223 [13], the multiplex used by the H.324

modem-based videophone standard [14]. It was investigated whether

compatibility could be achieved with this specification, which will

be used in future videophone-enabled (H.324 capable) modems.

However, since the multiplexing capabilities of H.223 are limited to

15 schedules (definitions of sequences of packet types that can be

identified in a multiplex header), for general Internet usage a

superset or a more general encapsulation would have been required.

Also, a PPP-style negotiation protocol was needed instead of using

(and necessarily extending) ITU-T H.245 [15] for setting the

parameters of the multiplex. In the PPP context, the interactions

with the encapsulations for data compression and link layer

encryption needed to be defined (including operation in the presence

of padding). But most important, H.223 requires synchronous HDLC

chips that can be configured to send frames without an attached CRC,

which is not possible with all chips deployed in commercially

available routers; so complete compatibility was unachievable.

Instead of adopting H.223, it was decided to pursue an approach that

is oriented towards compatibility both with existing hardware and

existing software (in particular PPP) implementations. The next

subsection groups these implementations according to their

capabilities.

4.3. Implementation models

This section introduces a number of terms for types of

implementations that are likely to emerge. It is important to have

these different implementation models in mind as there is no single

approach that fits all models best.

4.3.1. Sender types

There are two fundamental approaches to real-time transmission on

low-bitrate links:

Sender type 1

The PPP real-time framing implementation is able to control the

transmission of each byte being transmitted with some known,

bounded delay (e.g., due to FIFOs). For example, this is

generally true of PC host implementations, which directly access

serial interface chips byte by byte or by filling a very small

FIFO. For type 1 senders, a suspend/resume type approach will be

typically used: When a long frame is to be sent, the attempt is to

send it undivided; only if higher priority packets come up during

the transmission will the lower-priority long frame be suspended

and later resumed. This approach allows the minimum variation in

access delay for high-priority packets; also, fragmentation

overhead is only incurred when actually needed.

Sender type 2

With type 2 senders, the interface between the PPP real-time

framing implementation and the transmission hardware is not in

terms of streams of bytes, but in terms of frames, e.g., in the

form of multiple (prioritized) send queues directly supported by

hardware. This is often true of router systems for synchronous

links, in particular those that have to support a large number of

low-bitrate links. As type 2 senders have no way to suspend a

frame once it has been handed down for transmission, they

typically will use a queues-of-fragments approach, where long

packets are always split into units that are small enough to

maintain the access delay goals for higher-priority traffic.

There is a trade-off between the variation in access delay

resulting from a large fragment size and the overhead that is

incurred for every long packet by choosing a small fragment size.

4.3.2. Receiver types

Although the actual work of formulating transmission streams for

real-time applications is performed at the sender, the ability of the

receiver to immediately make use of the information received depends

on its characteristics:

Receiver type 1

Type 1 receivers have full control over the stream of bytes

received within PPP frames, i.e., bytes received are available

immediately to the PPP real-time framing implementation (with some

known, bounded delay e.g. due to FIFOs etc.).

Receiver type 2

With type 2 receivers, the PPP real-time framing implementation

only gets hold of a frame when it has been received completely,

i.e., the final flag has been processed (typically by some HDLC

chip that directly fills a memory buffer).

4.4. Conclusion

As a result of the diversity in capabilities of current

implementations, there are now two specifications for real-time

encapsulation: One, the multi-class extension to the PPP multi-link

protocol, is providing the solution for the queues-of-fragments

approach by extending the single-stream PPP multi-link protocol by

multiple classes [8]. The other encapsulation, PPP in a real-time

oriented HDLC-like framing, builds on this specification end extends

it by a way to dynamically delimit multiple fragments within one HDLC

frame [9], providing the solution for the suspend/resume type

approach.

5. Principles of Header Compression for Real-Time Flows

A good baseline for a discussion about header compression is in the

new IP header compression specification that was designed in

conjunction with the development of IPv6 [2]. The techniques used

there can reduce the 28 bytes of IPv4/UDP header to about 6 bytes

(depending on the number of concurrent streams); with the remaining 4

bytes of HDLC/PPP overhead and 12 bytes for RTP the total header

overhead can be about halved but still exceeds the size of a G.723.1

ACELP frame. Note that, in contrast to IP header compression, the

environment discussed here assumes the existence of a full-duplex PPP

link and thus can rely on negotiation where IP header compression

requires repeated transmission of the same information. (The use of

the architecture of the present document with link layer multicasting

has not yet been examined.)

Additional design effort was required for RTP header compression.

Applying the concepts of IP header compression, of the (at least) 12

bytes in an RTP header, 7 bytes (timestamp, sequence, and marker bit)

would qualify as RANDOM; DELTA encoding cannot generally be used

without further information since the lower layer header does not

unambiguously identify the semantics and there is no TCP checksum

that can be relied on to detect incorrect decompression. Only a more

semantics-oriented approach can provide better compression (just as

RFC1144 can provide very good compression of TCP headers by making

use of semantic knowledge of TCP and its checksumming method).

For RTP packets, differential encoding of the sequence number and

timestamps is an efficient approach for certain cases of payload data

formats. E.g., speech flows generally have sequence numbers and

timestamp fields that increase by 1 and by the frame size in

timestamp units, resp.; the CRTP (compressed RTP) specification makes

use of this relationship by encoding these fields only when the

second order difference is non-zero [7].

6. Announcement Protocols Used by Applications

As argued, the compressor can operate best if it can make use of

information that clearly identifies real-time streams and provides

information about the payload data format in use.

If these systems are routers, this consent must be installed as

router state; if these systems are hosts, it must be known to their

networking kernels. Sources of real-time information flows are

already describing characteristics of these flows to their kernels

and to the routers in the form of TSpecs in RSVP PATH messages [4].

Since these messages make use of the router alert option, they are

seen by all routers on the path; path state about the packet stream

is normally installed at each of these routers that implement RSVP.

Additional RSVP objects could be defined that are included in PATH

messages by those applications that desire good performance over low-

bitrate links; these objects would be coded to be ignored by routers

that are not interested in them (class number 11bbbbbb as defined in

[4], section 3.10).

Note that the path state is available in the routers even when no

reservation is made; this allows informed compression of best-effort

traffic. It is not quite clear, though, how path state could be torn

down quickly when a source ceases to transmit.

7. Elements of Hop-By-Hop Negotiation Protocols

The IP header compression specification attempts to account for

simplex and multicast links by providing information about the

compressed streams only in the forward direction. E.g., a full

IP/UDP header must be sent after F_MAX_TIME (currently 3 seconds),

which is a negligible total overhead (e.g. one full header every 150

G.723.1 packets), but must be considered carefully in scheduling the

real-time transmissions. Both simplex and multicast links are not

prevailing in the low-bitrate environment (although multicast

functionality may become more important with wireless systems); in

this document, we therefore assume full-duplex capability.

As compression techniques will improve, a negotiation between the two

peers on the link would provide the best flexibility in

implementation complexity and potential for extensibility. The peer

routers/hosts can decide which real-time packet streams are to be

compressed, which header fields are not to be sent at all, which

multiplexing information should be used on the link, and how the

remaining header fields should be encoded. PPP, a well-tried suite

of negotiation protocols, is already used on most of the low-bitrate

links and seems to provide the obvious approach. Cooperation from

PPP is also needed to negotiate the use of real-time encapsulations

between systems that are not configured to automatically do so.

Therefore, PPP options that can be negotiated at the link setup (LCP)

phase are included in [8], [9], and [10].

8. Security Considerations

Header compression protocols that make use of assumptions about

application protocols need to be carefully analyzed whether it is

possible to subvert other applications by maliciously or

inadvertently enabling their use.

It is generally not possible to do significant hop-by-hop header

compression on encrypted streams. With certain security policies, it

may be possible to run an encrypted tunnel to a network access server

that does header compression on the decapsulated packets and sends

them over an encrypted link encapsulation; see also the short mention

of interactions between real-time encapsulation and encryption in

section 4 above. If the security requirements permit, a special RTP

payload data format that encrypts only the data may preferably be

used.

9. References

[1] Handley, M., Crowcroft, J., Bormann, C. and J. Ott, "The

Internet Multimedia Conferencing Architecture", Work in

Progress.

[2] Degermark, M., Nordgren, B. and S. Pink, "IP Header

Compression", RFC2507, February 1999.

[3] Scott Petrack, Ed Ellesson, "Framework for C/RTP: Compressed

RTP Using Adaptive Differential Header Compression",

contribution to the mailing list rem-conf@es.net, February

1996.

[4] Braden, R., Zhang, L., Berson, S., Herzog, S. and S. Jamin,

"Resource ReSerVation Protocol (RSVP) -- Version 1 Functional

Specification", RFC2205, September 1997.

[5] Sklower, K., Lloyd, B., McGregor, G., Carr, D. and T.

Coradetti, "The PPP Multilink Protocol (MP)", RFC1990, August

1996.

[6] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,

"RTP: A Transport Protocol for Real-Time Applications", RFC

1889, January 1996.

[7] Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for

Low-Speed Serial Links", RFC2508, February 1999.

[8] Bormann, C., "The Multi-Class Extension to Multi-Link PPP", RFC

2686, September 1999.

[9] Bormann, C., "PPP in a Real-time Oriented HDLC-like Framing",

RFC2687, September 1999.

[10] Engan, M., Casner, S. and C. Bormann, "IP Header Compression

over PPP", RFC2509, February 1999.

[11] Wroclawski, J., "Specification of the Controlled-Load Network

Element Service", RFC2211, September 1997.

[12] Shenker, S., Partridge, C. and R. Guerin. "Specification of

Guaranteed Quality of Service", RFC2212, September 1997.

[13] ITU-T Recommendation H.223, "Multiplexing protocol for low bit

rate multimedia communication", International Telecommunication

Union, Telecommunication Standardization Sector (ITU-T), March

1996.

[14] ITU-T Recommendation H.324, "Terminal for low bit rate

multimedia communication", International Telecommunication

Union, Telecommunication Standardization Sector (ITU-T), March

1996.

[15] ITU-T Recommendation H.245, "Control protocol for multimedia

communication", International Telecommunication Union,

Telecommunication Standardization Sector (ITU-T), March 1996.

10. Author's Address

Carsten Bormann

Universitaet Bremen FB3 TZI

Postfach 330440

D-28334 Bremen, GERMANY

Phone: +49.421.218-7024

Fax: +49.421.218-7000

EMail: cabo@tzi.org

Acknowledgements

Much of the early discussion that led to this document was done with

Scott Petrack and Cary Fitzgerald. Steve Casner, Mikael Degermark,

Steve Jackowski, Dave Oran, the other members of the ISSLL subgroup

on low bitrate links (ISSLOW), and in particular the ISSLL WG co-

chairs Eric Crawley and John Wroclawski have helped in making this

architecture a reality.

Full Copyright Statement

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.

Acknowledgement

Funding for the RFCEditor function is currently provided by the

Internet Society.