





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.

   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
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   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
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   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.



















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