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Versions: (draft-camarillo-sip-compression) 00 01 02 RFC 3486

Internet Engineering Task Force                                   SIP WG
Internet Draft                                              G. Camarillo
                                                                Ericsson
draft-ietf-sip-compression-01.txt
September 17, 2002
Expires: March 2003


              Compressing the Session Initiation Protocol

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.


Abstract

   This document describes a mechanism to signal that compression is
   desired for one or more SIP messages. It also states when it is
   appropriate to send compressed SIP messages to a SIP entity.














G. Camarillo                                                  [Page 1]

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                           Table of Contents



   1          Introduction ........................................    3
   2          Overview of operation ...............................    4
   3          SigComp implementations for SIP .....................    4
   4          Sending a Request to a Server .......................    4
   4.1        Obtaining a SIP or SIPS URI with comp=sigcomp .......    5
   5          Sending a Response to a Client ......................    6
   6          Error Situations ....................................    7
   7          Augmented BNF .......................................    7
   8          Example .............................................    7
   9          Security Considerations .............................   10
   10         Acknowledges ........................................   10
   11         Authors' Addresses ..................................   10
   12         Normative References ................................   11
   13         Informative References ..............................   11






























G. Camarillo                                                  [Page 2]

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1 Introduction

   A SIP [1] client sending a request to a SIP server typically performs
   a DNS lookup for the domain name of the server. When NAPTR [3] or SRV
   [4] records are available for the server, the client can specify the
   type of service it wants. The service in this context is the
   transport protocol to be used by SIP (e.g., UDP, TCP or SCTP). A SIP
   server that support, for instance, three different transport
   protocols, will have three different DNS entries.

   Since it is foreseen that the number of transport protocols supported
   by a particular application layer protocol is not going to grow
   dramatically, having a DNS entry per transport seems like a scalable
   enough solution.

   However, sometimes it is necessary to include new layers between the
   transport protocol and the application layer protocol. Examples of
   these layers are transport layer security and compression. If DNS was
   used to discover the availability of these layers for a particular
   server, the number of DNS entries needed for that server would grow
   dramatically.

   A server that, for example, supported TCP and SCTP as transports, TLS
   for transport security and SigComp for signaling compression, would
   need the 8 DNS entries listed below:

        1.   TCP, no security, no compression

        2.   TCP, no security, SigComp

        3.   TCP, TLS, no compression

        4.   TCP, TLS, SigComp

        5.   SCTP, no security, no compression

        6.   SCTP, no security, SigComp

        7.   SCTP, TLS, no compression

        8.   SCTP, TLS, SigComp

   It is clear that this way of using DNS is not scalable. Therefore, an
   application layer mechanism to express support of signalling
   compression is needed.

        Note that for historical reasons both HTTP and SIP use a
        different port for TLS on top of TCP than for TCP alone,



G. Camarillo                                                  [Page 3]

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        although at present, this solution is not considered
        scalable any longer.

   A SIP element that supports compression will need to be prepared to
   receive compressed and uncompressed messages on the same port. It
   will perform demultiplexing based on the cookie in the topmost bits
   of every compressed message.

2 Overview of operation

   There are two types of SIP messages; SIP requests and SIP responses.
   Clients send SIP requests to the host part of a URI and servers send
   responses to the host in the sent-by parameter of the Via header
   field.

   We define two parameters, one for SIP URIs and the other for the Via
   header field. The format of both parameters is the same, as shown in
   the examples below:

   sip:alice@atlanta.com;comp=sigcomp
   Via: SIP/2.0/UDP server1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp


   The presence of this parameter (comp=sigcomp) in a URI indicates that
   the request has to be compressed using SigComp, as defined in [2].
   The presence of comp=sigcomp in a Via header field indicates that the
   response has to be compressed using SigComp.

   Therefore, the presence of comp=sigcomp indicates that the SIP entity
   identified by the URI or by the Via header field supports SigComp and
   is willing to receive compressed messages. Having comp=sigcomp mean
   "willingness" as well as "support" allows the receiver of a SIP
   message to influence the decision of whether or not to use SigComp at
   a given time.

3 SigComp implementations for SIP

   Note that, as stated in [5], every implementation that supports
   SigComp needs to implement the static dictionary for SIP and SDP.

   In addition, every SIP implementation that supports SigComp MUST
   implement the procedures described in this document.

4 Sending a Request to a Server

   A request is sent to the host part of a URI. This URI, referred to as
   the next-hop URI, is the Request-URI of the request or an entry in
   the Route header field.



G. Camarillo                                                  [Page 4]

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   If the next-hop URI contains the parameter comp=sigcomp, the client
   SHOULD compress the request using SigComp as defined in [2].

   If the next-hop URI is a SIPS URI, the request MUST be compressed
   before it is passed to the TLS layer.

   A client MUST NOT send a compressed request to a server if it does
   not know whether or not the server supports SigComp.

   Regardless of whether the request is sent compressed or not, if a
   client would like to receive subsequent requests within the same
   dialog in the UAS->UAC direction compressed, this client SHOULD add
   the parameter comp=sigcomp to the URI in the Contact header field if
   it is a user agent client. If the client is a proxy, it SHOULD add
   the parameter comp=sigcomp to its URI in the Record-Route header
   field.

   If a user agent client sends a compressed request, it SHOULD add the
   parameter comp=sigcomp to the URI in the Contact header field. If a
   proxy that Record-Routes sends a compressed request, it SHOULD add
   comp=sigcomp to its URI in the Record-Route header field.

   If a client sends a compressed request, it SHOULD add the parameter
   comp=sigcomp to the topmost entry of the Via header field.

   If a client does not know whether or not the server supports SigComp,
   but in case the server supported it, it would like to receive
   compressed responses, this client SHOULD add the parameter
   comp=sigcomp to the topmost entry of the Via header field. The
   request, however, as stated above, will not be compressed.

4.1 Obtaining a SIP or SIPS URI with comp=sigcomp

   For requests within a dialog, a next-hop URI with the comp=sigcomp
   parameter is obtained from a Record-Route header field when the
   dialog is established. A client sending a request outside a dialog
   can also obtain SIP URIs with comp=sigcomp in a Contact header field
   in a 3xx or 485 response to the request.

   However, clients establishing a session will not typically be willing
   to wait until the dialog is established in order to begin compressing
   messages. One of the biggest gains that SigComp can bring to SIP is
   the ability to compress the initial INVITE of a dialog, when the user
   is waiting for the session to be established. Therefore, clients need
   a means to obtain a comp=sigcomp URI from their outbound proxy before
   the user decides to establish a session.

   One solution to this problem is manual configuration. However,



G. Camarillo                                                  [Page 5]

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   sometimes it is necessary to have clients configured in an automatic
   fashion. Unfortunately, current mechanisms for SIP client
   configuration (e.g., using DHCP [6]) do not allow to provide the
   client with URI parameters. In this case, the client SHOULD send an
   uncompressed OPTIONS request to its outbound proxy. The outbound
   proxy can provide an alternative SIP URI with the comp=sigcomp
   parameter in a Contact header field in a 200 OK response to the
   OPTIONS. The client can use this URI for subsequent requests that are
   sent through the same outbound proxy using compression.

   RFC3261 [1] does not define how a proxy should respond to an OPTIONS
   request addressed to itself. It only describes how servers respond to
   OPTIONS addressed to a particular user. Section 11.2 of RFC3261 says:

        Contact header fields MAY be present in a 200 (OK) response
        and have the same semantics as in a 3xx response. That is,
        they may list a set of alternative names and methods of
        reaching the user.

   We extend this behavior to proxy servers responding to OPTIONS
   addressed to them. They MAY list a set of alternative URIs to contact
   the proxy.

   Note that receiving incoming requests (even initial INVITEs)
   compressed is not a problem, since user agents can REGISTER a SIP URI
   with comp=sigcomp in their registrar. All incoming requests for the
   user will be sent to this SIP URI using compression.

5 Sending a Response to a Client

   A response is sent to the host in the sent-by parameter of the Via
   header field. If the topmost Via header field contains the parameter
   comp=sigcomp, the response SHOULD be compressed. Otherwise, the
   response MUST NOT be compressed.

   A proxy performing Record-Route inspects the Record-Route header
   field in the response and the Contact header field in the request
   that triggered this response (see example in Section 8). It looks for
   the URI of the next upstream (closer to the user agent client) hop in
   the route set. If this URI contains the parameter comp=sigcomp, the
   proxy SHOULD add comp=sigcomp to its entry in the Record-Route header
   field. If this URI does not contain the parameter comp=sigcomp, the
   proxy SHOULD remove comp=sigcomp (if it is present) from its entry in
   the Record-Route header field.

   The same way, a user agent server SHOULD add comp=sigcomp to the
   Contact header field of the response if the URI of the next upstream
   hop in the route set contained the parameter comp=sigcomp.



G. Camarillo                                                  [Page 6]

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6 Error Situations

   If a compressed SIP request arrives to a SIP server that does not
   understand SigComp, the server will not have any means to indicate
   the error to the client. The message will be impossible to parse, and
   there will be no Via header field indicating an address to send the
   error response.

   If a SIP client sends a compressed request and the client transaction
   times out without having received any response, the client SHOULD
   retry the same request without using compression. If the compressed
   request was sent over a TCP connection, the client SHOULD close that
   connection and open a new one to send the uncompressed request.
   Otherwise the server would not be able to detect the beginning of the
   new message.

7 Augmented BNF

   This section provides the augmented Backus-Naur Form (BNF) of both
   parameters described above.

   The compression URI parameter is a "uri-parameter", as defined by the
   SIP ABNF (Section 25.1 of [1]):


        compression-param  =  "comp=" ("sigcomp" / other-compression)
        other-compression  =  token


   The Via compression parameter is a "via-extension", as defined by the
   SIP ABNF (Section 25.1 of [1]):


        via-compression    =  "comp" EQUAL ("sigcomp" / other-compression)
        other-compression  =  token


8 Example

   The following example illustrates the use of the parameters defined
   above. The call flow of Figure 1 shows an INVITE-200 OK-ACK handshake
   between a UAC and a UAS through two proxies. Proxy P1 does not
   Record-Route but proxy P2 does. Both proxies support compression, but
   they do not use it by default.

    UAC            P1            P2           UAS

     |(1)INVITE(c) |             |             |



G. Camarillo                                                  [Page 7]

Internet Draft                    SIP                 September 17, 2002


     |------------>| (2) INVITE  |             |
     |             |------------>| (3) INVITE  |
     |             |             |------------>|
     |             |             | (4) 200 OK  |
     |             | (5) 200 OK  |<------------|
     |(6)200 OK(c) |<------------|             |
     |<------------|             |             |
     |             |  (7)ACK(c)  |             |
     |-------------------------->|   (8) ACK   |
     |             |             |------------>|
     |             |             |             |
     |             |             |             |



   Figure 1: INVITE transaction through two proxies

   Messages (1), (6) and (7) are compressed (c).

   We provide a partial description of the messages involved in this
   call flow below. Only some parts of each message are shown, namely
   the Method name, the Request-URI and the Via, Route, Record-Route and
   Contact header fields. We have not used a correct format for these
   header fields. We have rather focus on the contents of the header
   fields and on the presence (or absence) of the "comp=sigcomp"
   parameter.

   (1) INVITE UAS
       Via: UAC;comp=sigcomp
       Route:P1;comp=sigcomp
       Contact: UAC;comp=sigcomp


   P1 is the outbound proxy of the UAC, and it supports SigComp. The UAC
   is configured to send compressed traffic to P1, and therefore, it
   compresses the INVITE (1). In addition, the UAC wants to receive
   future requests and responses for this dialog compressed. Therefore,
   it adds the comp=Sigcomp parameter to the Via and to the Contact
   header fields.

   (2) INVITE UAS
       Via: P1
       Via: UAC;comp=sigcomp
       Route:P2
       Contact: UAC;comp=sigcomp


   P1 forwards the INVITE (2) to P2. P1 does not use compression by



G. Camarillo                                                  [Page 8]

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   default, so it sends the INVITE uncompressed to P2.

   (3) INVITE UAS
       Via: P2
       Via: P1
       Via: UAC;comp=sigcomp
       Record-Route:P2
       Contact: UAC;comp=sigcomp


   P2 forwards the INVITE (3) to the UAS. P2 supports compression, but
   it does not use it by default. Therefore, it sends the INVITE
   uncompressed. P2 wishes to remain in the signalling path and
   therefore it Record-Routes.

   (4) 200 OK
       Via: P2
       Via: P1
       Via: UAC;comp=sigcomp
       Record-Route:P2
       Contact: UAS


   The UAS generates a 200 OK response and sends it to the host in the
   topmost Via, which is P2.

   (5) 200 OK
       Via: P1
       Via: UAC;comp=sigcomp
       Record-Route:P2;comp=sigcomp
       Contact: UAS


   P2 receives the 200 OK response. P2 Record-Routed, so it inspects the
   Route set for this dialog. For requests from the UAS towards the UAC
   (the opposite direction than the first INVITE), the next hop will be
   the Contact header field of the INVITE, because P1 did not Record-
   Route. That Contact identified the UAC:

       Contact: UAC;comp=sigcomp


   Since the UAC wants to receive compressed requests (Contact of the
   INVITE), P2 assumes that the UAC would also like to send compressed
   requests (Record-Route of the 200 OK). Therefore, P2 modifies its
   entry in the Record-Route header field of the 200 OK (5). In the
   INVITE (3), P2 did not used the comp=sigcomp parameter. Now it adds
   it in the 200 OK (5). This will allow the UAC sending compressed



G. Camarillo                                                  [Page 9]

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   requests within this dialog.

   (6) 200 OK
       Via: UAC;comp=sigcomp
       Record-Route:P2;comp=sigcomp
       Contact: UAS


   P1 sends the 200 OK (6) compressed to the UAC because the Via header
   field contained the comp=sigcomp parameter.

   (7) ACK UAS
       Via: UAC;comp=sigcomp
       Route:P2;comp=sigcomp
       Contact: UAC;comp=sigcomp


   The UAC sends the ACK (7) compressed directly to P2 (P1 did not
   Record-Route).

   (8) ACK UAS
       Via: P2
       Via: UAC;comp=sigcomp
       Contact: UAC;comp=sigcomp


   P2 sends the ACK (8) uncompressed to the UAS.

9 Security Considerations

   A SIP entity receiving a compressed message has to decompress it and
   to parse it. This requires slightly more processing power than only
   parsing a message. This implies that a denial of service attack using
   compressed messages would be slightly worse than an attack with
   uncompressed messages.

   An attacker inserting the parameter comp=sigcomp in a SIP message
   could make a SIP entity send compressed messages to another SIP
   entity that did not support SigComp. Appropriate integrity mechanisms
   should be used to avoid this attack.

10 Acknowledges

   Allison Mankin, Jonathan Rosenberg and Miguel Angel Garcia provided
   valuable comments on this memo.

11 Authors' Addresses




G. Camarillo                                                 [Page 10]

Internet Draft                    SIP                 September 17, 2002


   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland
   electronic mail:  Gonzalo.Camarillo@ericsson.com

12 Normative References

   [1] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.
   Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
   initiation protocol," RFC 3261, Internet Engineering Task Force, June
   2002.

   [2] R. Price  et al.  , "Signaling compression," Internet Draft,
   Internet Engineering Task Force, June 2002.  Work in progress.

13 Informative References

   [3] M. Mealling and R. Daniel, "The naming authority pointer (NAPTR)
   DNS resource record," RFC 2915, Internet Engineering Task Force,
   Sept. 2000.

   [4] A. Gulbrandsen, P. Vixie, and L. Esibov, "A DNS RR for specifying
   the location of services (DNS SRV)," RFC 2782, Internet Engineering
   Task Force, Feb. 2000.

   [5] M. Garcia  et al.  , "The session initiation protocol (SIP) and
   session description protocol (SDP) static dictionary for signaling
   compression (sigcomp)," Internet Draft, Internet Engineering Task
   Force, July 2002.  Work in progress.

   [6] H. Schulzrinne, "DHCP option for SIP servers," Internet Draft,
   Internet Engineering Task Force, Mar. 2002.  Work in progress.


   Full Copyright Statement

   Copyright (c) The Internet Society (2002). 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



G. Camarillo                                                 [Page 11]

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   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.




































G. Camarillo                                                 [Page 12]


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