Network Working Group H. Hannu
Request for Comments: 3322 Ericsson
Category: Informational January 2003
Signaling Compression (SigComp) Requirements & Assumptions
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.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
The purpose of this document is to outline requirements and
motivations for the development of a scheme for compression and
decompression of messages from signaling protocols. In wireless
environments and especially in cellular systems, e.g., GSM (Global
System for Mobile communications) and UMTS (Universal Mobile
Telecommunications System), there is a need to maximize the transport
efficiency for data over the radio interface. With the introduction
of SIP/SDP (Session Initiation Protocol/Session Description Protocol)
to cellular devices, compression of the signaling messages should be
considered in order to improve both service availability and quality,
mainly by reducing the user idle time, e.g., at call setup.
Table of Contents
1. Introduction....................................................2
1.1. Protocol Characteristics......................................2
1.2. Cellular System Radio Characteristics.........................3
2. Motivation for Signaling Reduction..............................4
2.1. Estimation of Call Setup Delay Using SIP/SDP..................4
3. Alternatives for Signaling Reduction............................6
4. Assumptions.....................................................7
5. Requirements....................................................8
5.1. General Requirements..........................................8
5.2. Performance Requirements......................................9
6. Security Considerations.........................................11
7. IANA Considerations.............................................11
8. References......................................................11
9. Author's Address................................................12
10. Full Copyright Statement.......................................13
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1. Introduction
In wireless environments, and especially in cellular systems, such as
GSM/GPRS, there is a need to maximize the transport efficiency of
data over the radio interface. The radio spectrum is rather
expensive and must be carefully used. Therefore, the cellular
systems must support a sufficient number of users to make them
economically feasible. Thus, there is a limitation in the per user
bandwidth.
Compressing the headers of the network and transport protocols used
for carrying user data is one way to make more efficient use of the
scarce radio resources [ROHC]. However, compression of the messages
from signaling protocols, such as SIP/SDP, should also be considered
to increase the radio resource usage even further. Compression will
also improve the service quality by reducing the user idle time at
e.g., call setup. When IP is used end-to-end, new applications, such
as streaming, will be brought to tiny end-hosts, such as cellular
devices. This will introduce additional traffic in cellular systems.
Compression of signaling messages, such as RTSP [RTSP], should also
be considered to improve both the service availability and quality.
New services with their corresponding signaling protocols make it
reasonable to consider a scheme that is generic. The scheme should
be generic in the meaning that the scheme can efficiently be applied
to arbitrary protocols with certain characteristics, such as the
ASCII based protocols SIP and RTSP.
1.1. Protocol Characteristics
The following application signaling protocols are examples of
protocols that are expected to be commonly used in the future. Some
of their characteristics are described below.
1.1.1 SIP
The Session Initiation Protocol [SIP] is an application layer
protocol for establishing, modifying and terminating multimedia
sessions or calls. These sessions include Internet multimedia
conferences, Internet telephony and similar applications. SIP can be
used over either TCP [TCP] or UDP [UDP]. SIP is a text based
protocol, using ISO 10646 in UTF-8 encoding.
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1.1.2 SDP
The Session Description Protocol [SDP] is used to advertise
multimedia conferences and communicate conference addresses and
conference tool specific information. It is also used for general
real-time multimedia session description purposes. SDP is carried in
the message body of SIP and RTSP messages. SDP is text based using
the ISO 10646 character set in UTF-8 encoding.
1.1.3 RTSP
The Real Time Streaming Protocol [RTSP] is an application level
protocol for controlling the delivery of data with real-time
properties, such as audio and video. RTSP may use UDP or TCP (or
other) as a transport protocol. RTSP is text based using the ISO
10646 character set in UTF-8 encoding.
1.1.4 Protocol Similarities
The above protocols have many similarities. These similarities will
have implications on solutions to the problems they create in
conjunction with e.g., cellular radio access. The similarities
include:
- Requests and reply characteristics. When a sender sends a
request, it stays idle until it has received a response. Hence,
it typically takes a number of round trip times to conclude e.g.,
a SIP session.
- They are ASCII based.
- They are generous in size in order to provide the necessary
information to the session participants.
- SIP and RTSP share many common header field names, methods and
status codes. The traffic patterns are also similar. The
signaling is carried out primarily under the set up phase. For
SIP, this means that the majority of the signaling is carried out
to set up a phone call or multimedia session. For RTSP, the
majority of the signaling is done before the transmission of
application data.
1.2. Cellular System Radio Characteristics
Partly to enable high utilization of cellular systems, and partly due
to the unreliable nature of the radio media, cellular links have
characteristics that differ somewhat from a typical fixed link, e.g.,
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copper or fiber. The most important characteristics are the lossy
behavior of cellular links and the large round trip times.
The quality in a radio system typically changes from one radio frame
to another due to fading in the radio channel. Due to the nature of
the radio media and interference from other radio users, the average
bit error rate (BER) can be 10e-3 with a variation roughly between
10e-2 to 10e-4. To be able to use the radio media with its error
characteristics, methods such as forward error correction (FEC) and
interleaving are used. If these methods were not used, the BER of a
cellular radio channel would be around 10 %. Thus, radio links are,
by nature, error prone. The final packet loss rate may be further
reduced by applying low level retransmissions (ARQ) over the radio
channel; however, this trades decreased packet loss rate for a larger
delay. By applying methods to decrease BER, the system delay is
increased. In some cellular systems, the algorithmic channel round
trip delay is in the order of 80 ms. Other sources of delays are
DSP-processing, node-internal delay and transmission. A general
value for the RTT is difficult to state, but it might be as high as
200 ms.
For cellular systems it is of vital importance to have a sufficient
number of users per cell; otherwise the system cost would prohibit
deployment. It is crucial to use the existing bandwidth carefully;
hence the average user bit rate is typically relatively low compared
to the average user bit rate in wired line systems. This is
especially important for mass market services like voice.
2. Motivation for Signaling Reduction
The need for solving the problems caused by the signaling protocol
messages is exemplified in this chapter by looking at a typical
SIP/SDP Call Setup sequence over a narrow band channel.
2.1 Estimation of Call Setup Delay Using SIP/SDP
Figure 2.1 shows an example of SIP signaling between two termination
points with a wireless link between, and the resulting delay under
certain system assumptions.
It should be noted that the used figures represent a very narrow band
link. E.g., a WCDMA system can provide maximum bit rates up to 2
Mbits/s in ideal conditions, but that means one single user would
consume all radio resources in the cell. For a mass market service
such as voice, it is always crucial to reduce the bandwidth
requirements for each user.
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Client Network-Proxy Size [bytes] Time [ms]
| |
|---------- INVITE --------->| 620 517+70=587
| |
|<-- 183 Session progress ---| 500 417+70=487
| |
|---------- PRACK ---------->| 250 208+70=278
| |
|<----- 200 OK (PRACK) ------| 300 250+70=320
: :
|<...... RSVP and SM .......>|
: :
|---------- COMET ---------->| 620 517+70=587
| |
|<----- 200 OK (COMET) ------| 450
| | +
|<------ 180 Ringing --------| 230 567+70=637
| |
|---------- PRACK ---------->| 250 208+70=278
| |
|<----- 200 OK (PRACK) ------| 300
| | +
|<--------- 200 OK ----------| 450 625+70=695
| |
|----------- ACK ----------->| 230 192+70=262
Figure 2.1. SIP signaling delays assuming a link speed of 9600
bits/sec and a RTT of 140 ms.
The one way delay is calculated according to the following equation:
OneWayDelay =
MessageSize[bits]/LinkSpeed[bits/sec] + RTT[sec]/2 (eq. 1)
The following values have been used:
RTT/2: 70 ms
LinkSpeed 9.6 kbps
The delay formula is based on an approximation of a WCDMA radio
access method for speech services. The approximation is rather
crude. For instance, delays caused by possible retransmissions due
to errors are ignored. Further, these calculations also assume that
there is only one cellular link in the path and take delays in an
eventual intermediate IP-network into account. Even if this
approximation is crude, it is still sufficient to provide
representative numbers and enable comparisons. The message size
given in Figure 2.1, is typical for a SIP/SDP call setup sequence.
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2.1.1 Delay Results
Applying equation 1 to each SIP/SDP message shown in the example of
Figure 2.1 gives a total delay of 4131 ms from the first SIP/SDP
message to the last. The RSVP and Session Management (Radio Bearer
setup), displayed in Figure 2.1, will add approximately 1.5 seconds
to the total delay, using equation 1. However, there will also be
RSVP and SM signaling prior to the SIP INVITE message to establish
the radio bearer, which would add approximately another 1.5 seconds.
In [TSG] there is a comparison between GERAN call setup using SIP and
ordinary GSM call setup. For a typical GSM call setup, the time is
about 3.6 seconds, and for the case when using SIP, the call setup is
approximately 7.9 seconds.
Another situation that would benefit from reduced signaling is
carrying signaling messages over narrow bandwidth links in mid-call.
For GERAN, this will result in frame stealing with degraded speech
quality as a result.
Thus, solutions are needed to reduce the signaling delay and the
required bandwidth when considering both system bandwidth
requirements and service setup delays.
3. Alternatives for Signaling Reduction
More or less attractive solutions to the previously mentioned
problems can be outlined:
- Increase the user bit rate
An increase of the bit rate per user will decrease the number of
users per cell. There exist systems (for example WCDMA) which can
provide high bit rates and even variable rates, e.g., at the setup
of new sessions. However, there are also systems, e.g., GSM/EDGE,
where it is not possible to reach these high bit rates in all
situations. At the cell borders, for example, the signal strength
to noise ratio will be lower and result in a lower bit rate. In
general, an unnecessary increase of the bit rate should be avoided
due to the higher system cost introduced and the possibility of
denial of service. The latter could, for example, be caused by
lack of enough bandwidth to support the sending of the large setup
message within a required time period, which is set for QoS
reasons.
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- Decrease the RTT of the cellular link
Decreasing the RTT would require substantial system changes and is
thus not feasible in the short term. Further, the RTT-delay
caused by interleaving and FEC will always have to be present
regardless of which system is used. Otherwise the BER will be too
high for the received data to be useful, or alternatively trigger
retransmissions giving an average total delay of the same or
higher magnitude.
- Optimize message sequence for the protocols
If the request/response pattern could be eased up, then "keeping
the pipe full" could be a way forward. Thus, instead of following
the message sequence described in Figure 4.2, more than one
message would be sent in a row, even though no response has been
received. However, this would entail protocol changes and may be
difficult at the current date.
- Protocol stripping
Removing fields from a message would decrease the size of the
messages to some extent. However, this would cause the loss of
transparency and thus violate the End-to-End principle and is thus
not desirable.
- Compression
By compressing messages, the impact of the mentioned problems
could be decreased. Compared to the other possible solutions
compression can be made, and must be, transparent to the end-user
application. Thus, compression seems to be the most attractive
way forward.
4. Assumptions
- Negotiation
How the usage of compression is negotiated is out of the scope for
this compression solution and must be handled by e.g., the
protocol the messages of which are to be compressed.
- Reliable transport
With reliable transport, it is assumed that a transport recovered
from data that is damaged, lost, duplicated, or delivered out of
order, e.g., [TCP].
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- Unreliable transport
With unreliable transport, it is assumed that a transport does not
have the capabilities of a reliable transport, e.g., [UDP].
5. Requirements
This chapter states requirements for a signaling compression scheme
to be developed in the IETF ROHC WG.
The requirements are divided into two parts. Section 5.1 sets
general requirements concerning the Internet infrastructure, while
Section 5.2 sets requirements on the scheme itself.
5.1. General Requirements
1. Transparency: When a message is compressed and then decompressed,
the result must be bitwise identical to the original message.
Justification: This is to ensure that the compression scheme will
not cause problems for any current or future part of the Internet
infrastructure.
Note: See also requirement 9.
2. Header compression coexistence: The compression scheme must be
able to coexist with header compression, especially the ROHC
protocol.
Justification: Signaling compression is used because there is a
need to conserve bandwidth usage. In that case, header
compression will likely be needed too.
3a. Compatibility: The compression scheme must be constructed in such
a way that it allows the above protocols' mechanisms to negotiate
whether the compression scheme is to be applied or not.
Justification: Two entities must be able to communicate
regardless if the signaling compression scheme is implemented at
both entities or not.
3b. Ubiquity: Modifications to the protocols generating the messages
that are to be compressed, must not be required for the
compression scheme to work.
Justification: This will simplify deployment of the compression
scheme.
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Note: This does not preclude making extensions, which are related
to the signaling compression scheme, to existing protocols, as
long as the extensions are backward compatible.
4. Generality: Compression of arbitrary message streams must be
supported. The signaling compression scheme must not be limited
to certain protocols, traffic patterns or sessions. It must not
assume any message pattern to be able to perform compression.
Justification: There might be a future need for compression of
different ASCII based signaling protocols. This requirement will
minimize future work.
Note: This does not preclude optimization for certain streams.
5. Unidirectional routes: The compression scheme must be able to
operate on unidirectional routes, i.e., without explicit feedback
messages from the decompressor.
Note: Implementations on unidirectional routes might possibly
show a degraded performance compared to implementations on bi-
directional routes.
6. Transport: The solution must work for both unreliable and
reliable underlying transport protocols, e.g., UDP and TCP.
Justification: The protocols, which generate the messages that
are to be compressed, may use either an unreliable or a reliable
underlying transport.
Note: This should not be taken to mean that the same set of
solution mechanisms must be used over both unreliable and
reliable transport.
5.2. Performance Requirements
The performance requirements in this section and the following
subsections are valid for both unreliable and reliable underlying
transport.
7. Scalability: The scheme must be flexible to accommodate a range
of compressors/decompressors with varying memory and processor
capabilities.
Justification: A primary target for the signaling compression
scheme is cellular systems, where the mobile terminals have
varying capabilities.
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8. Delay: The signaling compression must not noticeably add to the
delay experienced by the end user.
Justification: Reduction of the user experienced delay is the
main purpose of signaling compression.
Note: This requirement is intended to prevent schemes that
achieve compression efficiency at the expense of delay, i.e.,
queuing of messages to improve the compression efficiency should
be avoided.
The following requirements are grouped into two subsections, a
robustness section and a compression efficiency section.
5.2.1. Robustness
The requirements in this section concern the issue of when compressed
messages should be correctly decompressed. The transparency
requirement (first requirement) covers the issue with faulty
decompressed messages.
9. Residual errors: The compression scheme must be resilient against
errors undetected by lower layers, i.e., the probability of
incorrect decompression caused by such undetected errors must be
low.
Justification: A primary target for the signaling compression
scheme is cellular systems, where undetected errors might be
introduced on the cellular link.
10. Error propagation: Propagation of errors due to signaling
compression should be kept at an absolute minimum. Loss or
damage to a single or several messages, between compressor and
decompressor should not prevent compression and decompression of
later messages.
Justification: Error propagation reduces resource utilization and
quality.
11. Delay: The compression scheme must be able to perform compression
and decompression of messages under all expected delay
conditions.
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5.2.2. Compression Efficiency
This section states requirements related to compression efficiency.
12. Message loss: Loss or damage to a single or several messages, on
the link between compressor and decompressor, should not prevent
the usage of later messages in the compression and decompression
process.
13. Moderate message misordering: The scheme should allow for the
correct decompression of messages, that have been moderately
misordered (1-2 messages) between compressor and decompressor.
The scheme should not prevent the usage of later messages in the
compression and decompression process.
Justification: Misordering is frequent on the Internet, and this
kind of misordering is common.
6. Security Considerations
A protocol specified to meet these requirements must be able to cope
with packets that have undergone security measures, such as
encryption, without adding any security risks. This document, by
itself however, does not add any security risks.
7. IANA Considerations
A protocol which meets these requirements may require the IANA to
assign various numbers. This document by itself however, does not
require any IANA involvement.
8. References
[ROHC] Bormann, C., Burmeister, C., Degermark, M., Fukushima, H.,
Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K.,
Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke,
T., Yoshimura, T. and H. Zheng, "RObust Header Compression
(ROHC): Framework and four profiles: RTP, UDP, ESP, and
uncompressed", RFC 3095, July 2001.
[RTSP] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
[SDP] Handley, H. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
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[SIP] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[UDP] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
1980.
[TCP] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
[TSG] Nortel Networks, "A Comparison Between GERAN Packet-Switched
Call Setup Using SIP and GSM Circuit-Switched Call Setup Using
RIL3-CC, RIL3-MM, RIL3-RR, and DTAP", 3GPP TSG GERAN #2, GP-
000508, 6-10 November 2000.
9. Author's Address
Hans Hannu
Box 920
Ericsson AB
SE-971 28 Lulea, Sweden
Phone: +46 920 20 21 84
EMail: hans.hannu@epl.ericsson.se
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10. Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
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Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
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