One document matched: draft-calhoun-sip-aaa-reqs-01.txt

Differences from draft-calhoun-sip-aaa-reqs-00.txt







INTERNET DRAFT                                           Henrik Basilier
Category: Informational                                   Ericsson, Inc.
Title: draft-calhoun-sip-aaa-reqs-01.txt                  Pat R. Calhoun
Date: November 2000                               Sun Microsystems, Inc.
                                                           Matt Holdrege
                                                                 ipVerse
                                                          Tony Johansson
                                                          Ericsson, Inc.
                                                             James Kempf
                                                  Sun Microsystems, Inc.



              AAA Requirements for IP Telephony/Multimedia



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

   The list of Internet-Draft Shadow Directories can be accessed at:

      http://www.ietf.org/shadow.html.

   This document is an individual contribution for consideration by the
   SIP Working Group of the Internet Engineering Task Force.  Comments
   should be submitted to the diameter@diameter.org mailing list.

   Distribution of this memo is unlimited.

   Copyright   (C) The Internet Society 2000.  All Rights Reserved.






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Abstract

   The AAA Working Group has been defining requirements for Network
   Access in Inter-Domain (roaming) networks, including requirements
   from the Mobile IP and NASREQ Working Groups. Some of these
   requirements were input from work done in the 3rd Generation wireless
   SDOs. These same SDO's have a need to tie their IP Intelligent
   Servers (e.g. Softswitches, Call Agents, SIP Servers, etc.)  to their
   AAA infrastructure. This specification defines some requirements for
   both the AAA (meaning a new extension to DIAMETER) and IP
   Telephony/Multimedia protocols (SIP, MEGACO, etc.)

   This Internet Draft is intended to stimulate discussions within the
   SIP Working Group, in order to come up with a set of requirements
   that could then be used to begin informative protocol design in the
   AAA Working Group.

Table of Contents

      1.0  Introduction
            1.1  Requirements language
      2.0  Network Architecture
            2.1  Registration
            2.2  Invitation
                  2.2.1  Invitation terminating in the mobile node
                  2.2.2  Invitation originating from the mobile node
      3.0  Requirements
      4.0  Security Considerations
      5.0  References
      6.0  Acknowledgements
      7.0  Authors' Addresses
      8.0  Full Copyright Statement



















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1.0  Introduction


   The AAA Working Group has been defining requirements for Network
   Access in Inter-Domain (roaming) networks, including requirements
   from the Mobile IP and NASREQ Working Groups. Some of these
   requirements were input from work done in the 3rd Generation wireless
   SDOs. These same SDO's have a need to tie their IP Intelligent
   Servers (e.g. Softswitches, Call Agents, SIP Servers, etc.)  to their
   AAA infrastructure. This specification defines some requirements for
   both the AAA (meaning a new extension to DIAMETER) and IP
   Telephony/Multimedia protocols (SIP, MEGACO, etc.)

   This Internet Draft is intended to stimulate discussions within the
   SIP Working Group, in order to come up with a set of requirements
   that could then be used to begin informative protocol design in the
   AAA Working Group.


1.1  Requirements language

   In this document, the key words "MAY", "MUST, "MUST NOT", "optional",
   "recommended", "SHOULD", and "SHOULD NOT", are to be interpreted as
   described in [13].


2.0  Network Architecture

   This section will contain some details on how the authors envision
   SIP will be used in 3G as well as other wired and wireless networks.
   We will detail how the registration procedure will occur.  We will
   also investigate how a SIP invitation will proceed. All examples in
   this version of the document assumes that the SIP clients always
   registers with their home network (home control).

   The authors wish to state that much of this work is still in progress
   in various other groups including the SIP WG and 3G SDOs, and is
   subject to change. The ideas described in this document do not
   represent a final agreement in any working group or SDOs, but does
   include ideas from well established positions in the related groups.
   This document will be updated when further SIP, AAA, and 3G
   requirements are received.

   This document is also intended as an informal method for IETF SIP
   experts to feed in their expertise into the 3G standardization
   efforts through comments on this Internet Draft.

   Although a driving force towards the integration of SIP and AAA is



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   the next generation wireless networks (3G), it is desirable that the
   architecture proposed be similar, if not identical, to the wireline
   architecture.


2.1  Registration

   In this section, we will provide an example of a user registering his
   device. The scenario described is one where the user is roaming in a
   foreign network, typically owned and operated by a different service
   provider. This is however seen as a superset of the scenario where
   the user is in his home network, which is therefore not explicitely
   described.

   It is a requirement that a user is not be tied to a specific SIP
   server. This is necessary for many reasons, including scaling and to
   minimize deployment issues when SIP servers are added to the network
   to relieve traffic load. In this document, the SIP server that has
   been either statically or dynamically assigned to serve a particular
   user is called the Anchor SIP Proxy. This document assumes that the
   Anchor SIP proxy is always assigned in the home network of the user.

   One other important requirement is the ability to have a SIP server
   Central Point of Contact (SCPC), acting as a SIP proxy/redirect.  The
   SCPC will use a location database function to proxy/redirect messages
   to the Anchor SIP proxy serving the particular user.  This SCPC will
   be used both for incoming SIP Sessions (SIP Invites originating in
   another network) as well as messages originating from roaming mobiles
   (accessing functions in their home network from a different
   domain/provider). The location database function is the same as
   described in [1].


   There could be several methods for a mobile node to find its SCPC or
   for any other node to find the SCPC Contact of a user. Although the
   exact methods to use is outside the scope of this document, we have
   assumed the use of Domain Name Service (DNS) throughout this
   document.













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                  Home Network
                   +--------+       +----------+
          +------->|xyz.com |<----->| xyz.com  |
          |        |  AAAH  |<--+   | Location |
          |     +--| server +-+ |   | Database |
          |     |  +--------+ | |   +----------+
          |     |             | |        ^
     4.AAA|3.AAA|        5.AAA| |2.AAA   |
       Rsp|  Req|          Rsp| |  Req   |
          |     v             v |        v
       +--+--------+ 6. SIP +---+---------+
       |   Anchor  |    Reg.|     SIP     |
       |Softswitch/|<-------+   Central   |
       |    SIP    |        |   Point of  |
       |   Proxy   +------->|   Contact   |
       +-----------+7.200 OK+--+----------+
                               |    ^
                               |    |
                       8.200 OK|    | 1. SIP Register
                               |    |
                               |    |
                               v    |
                            +-------+-+
                            |   SIP   |
                            | Client  | bob@xyz.com
                            +---------+

                        Figure 1: SIP Registration


   When the SIP client issues its SIP Register message to the SCPC
   within its home network, a AAA message is issued to the Home AAA
   Server (AAAH).  The AAA message includes the SIP Client's identity,
   the SIP message as opaque data, and other authorization and service
   specific information.  The AAAH uses the information to authenticate
   the SIP client, and to determine whether it authorizes the client to
   access the service.

   If the AAAH determines that its clock, and that of the client, is
   synchronized, as evidenced by the Timestamp header field, it MAY
   process the message with some assurances that the message is not a
   replay of an old message. However, the AAAH MAY decide that a
   challenge-response procedure is appropriate, and instruct the SIP
   proxy to send back a 401 (Unauthorized) response. The AAAH will
   generate the challenge in the response message that is sent back in
   to the client, which will have to re-register.





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                  Home Network
                   +--------+       +----------+
          +------->|xyz.com |<----->| xyz.com  |
          |        |  AAAH  |<--+   | Location |
          |     +--| server +-+ |   | Database |
          |     |  +--------+ | |   +----------+
          |     |             | |        ^
     4.AAA|3.AAA|     5.AAA   | |2.AAA   |
       Rsp|  Req|     Response| |  Req   |
          |     v             v |        v
       +--+--------+        +---+---------+
       |   Anchor  |        |     SIP     |
       |Softswitch/|        |   Central   |
       |    SIP    |        |   Point of  |
       |   Proxy   |        |   Contact   |
       +--+--------+        +----+--------+
          |     ^                |  ^
          |     |                |  |
          |     |                |  | 1. SIP Register
          |     |      6.Redirect|  |
          |     |                |  |
          |     |7.SIP           v  |
          |     |  Register +-------+-+
          |     +-----------+   SIP   |
          +---------------->| Client  | bob@xyz.com
            8.200 OK        +---------+
                 Figure 2: SIP Registration with redirect.


   The SCPC MAY select an Anchor SIP proxy for the user, in such a case
   it MUST pass the address of the selected Anchor to the AAAH.

   If access is granted, the AAAH must verify whether a static Anchor
   SIP Server was configured for the client or if one was selected by
   the SCPC.  Otherwise, the AAAH dynamically assigns an Anchor SIP
   Server to the SIP client, which will act as the client's server for
   the duration of the authorization period (determined by the AAAH).
   The AAAH MAY also determine that the centralized SIP Server (the one
   that generated the AAA message) should act as the client's server for
   the duration of the authorization period.

   If the AAAH determines that the SIP client does not have a security
   association with the assigned server, it MAY create a dynamic
   security association between the nodes, by distributing session keys
   to be used in all subsequent SIP messages. This is similar to the
   method described in [4].

   Dynamic establishment of security associations by the AAAH is



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   typically useful in scenarios where the entities do not have pre-
   established trust, and trust is mandatory in all SIP messages. It
   should be equally possible for the AAAH to return the relevant
   certificates to the entities, which could then establish a direct
   security association, via an out-of-band key management protocol,
   such as the Internet Key Exchange [9].

   The AAAH MAY push security information (e.g Session keys) along with
   other necessary information (e.g Service profile) to the assigned
   Anchor SIP proxy. It then responds back to the SCPC and passes the
   address of the assigned Anchor SIP proxy. The SCPC then proxies the
   registration message to the Anchor SIP proxy, which responds with a
   200 OK message.

   If security information and other necessary information (e.g. Service
   Profile) has not been pushed to the Anchor SIP proxy, the Anchor SIP
   proxy MAY pull this information from the AAAH. This sequence is not
   illustrated in any of the figures.

   In the event that the AAAH created session keys for the communicating
   SIP entities, the SIP server MUST include the session keys destined
   for the SIP client within the SIP response. The server then adds its
   own authentication information using the session keys it received
   from the AAAH.  The response is subsequently forwarded to the SIP
   client.

   A successful SIP registration MAY result in the generation of an
   accounting record by the client's anchor SIP server. The accounting
   record MAY include such information as the user's identity, the
   location of the registration, date and time, etc.

   Once the AAAH has sent the successful authorization message to the
   anchor SIP server, it MAY update its local location database. The
   database contains the identity of the SIP client, and the anchor SIP
   server. This database MAY be used by other SIP servers within the
   local administrative domain to identify a SIP client's assigned SIP
   server.

   Figure 2 shows an alternative scenario that MAY be supported. The
   only difference is that the SCPC after authentication issues a
   redirect to the mobile node, sending it the address of the Anchor SIP
   proxy.  The mobile node MAY upon reception of this redirect message
   redirect all subsequent messaging directly to the Anchor SIP proxy,
   including SIP Invite messages, thus bypassing the SCPC.

2.2  Invitation

   In this section, we will look at the details of a SIP invite message,



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   and how a call is setup. Although it cannot be forbidden, it is
   assumed that the SIP Servers do not share pre-existing security
   association, either implicitely or via the AAA infrastructure. This
   allows SIP sessions to be established from users that are not members
   of the AAA infrastructure.

2.2.1  Invitation terminating in the mobile node




                   Home Network
                   +--------+    +----------+    +----------+
                   |xyz.com |    | xyz.com  |    |  Naming  |
                   |  AAAH  |    | Location |    |  Server  |
                +->| server |    | Database |    | (e.g.DNS)|
                |  +--------+    +----------+    +----------+
                |                  ^              ^
                |   AAA            | Location     |
                |                  | Query        | xyz.com?
                v                  v              v
       +-----------+ Invite+-----------+ Invite +-----------+
       |   Anchor  |<------+    SIP    |<-------|  abc.com  |
       |Softswitch/|       |  Central  |        |           |
       |    SIP    +------>|  Point of |------->|   SIP     |
       |   Proxy   |200 OK |  Contact  |200 OK  |  Server   |
       +-------+---+       +-----------+        +---+-------+
           ^   |                                    |  ^
     200 OK|   |                              200 OK|  |
           |   | SIP                                |  | SIP
           |   | Invite                             |  | Invite
           |   v                                    v  |
        +--+------+                               +---------+
        |   SIP   |                               |   SIP   |
        | Client  |                               | Client  |
        +---------+                               +---------+
         bob@xyz.com                              joe@abc.com

          Figure 3: Mobile Node terminated SIP Session Initiation

   Figure 3 and Figure 4 provides an example of a user that wishes to
   establish a SIP session with bob@xyz.com.

   The SIP client of joe@abc.com issues the invite message to its local
   SIP Server (abc.com SIP Server).

   If the Anchor SIP proxy of bob@xyz.com is not known, the SIP Server
   SHOULD attempt to resolve the SIP Central Point of Contact (SCPC)



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   within the home network, perhaps using a mechanism such as DNS. The
   Invite message is forwarded to the SCPC, which finds the user's
   current anchor SIP server by sending a query to the Location
   Database. At that point, the SCPC MAY either forward the request
   directly to the Anchor SIP Server (Figure 3), or return a redirect
   message to the initiating SIP Server (Figure 4). In either case, the
   message is forwarded to the Anchor SIP Server.


                  Home Network
                   +--------+    +----------+    +----------+
                   |xyz.com |    | xyz.com  |    |  Naming  |
                   |  AAAH  |    | Location |    |  Server  |
                +->| server |    | Database |    | (e.g.DNS)|
                |  +--------+    +----------+    +----------+
                |                  ^              ^
                |   AAA            | Location     |
                |                  | Query        | xyz.com?
                v                  v              v
       +-----------+       +-----------+ Invite +-----------+
       |   Anchor  |       |    SIP    |<-------|  abc.com  |
       |Softswitch/|       |  Central  |        |           |
       |    SIP    |<-+    |  Point of |------->|    SIP    |
       |   Proxy   |  |    |  Contact  |Redirect|   Server  |
       +-----------+  |    +-----------+        +-----------+
             ^        |        Invite             |    ^
             |        +---------------------------+    |
             |  SIP                             SIP    |
             |  Invite                          Invite |
             v                                         |
        +---------+                               +---------+
        |   SIP   |                               |   SIP   |
        | Client  |                               | Client  |
        +---------+                               +---------+
        bob@xyz.com                               joe@abc.com

   Figure 4: Mobile node terminated SIP Session Initiation (alternative)


   Upon receipt of the SIP Invite message, the Anchor SIP Server MAY
   send an authorization request to the AAAH, in order to determine
   whether the session can be established. The authorization check by
   the AAAH MAY include other policy decisions, such as time of day,
   origination point of the call, etc. A successful response from the
   AAAH will result in the forwarding of the SIP Invite message to the
   SIP Client. A response that includes an error would cause the Anchor
   SIP Server to return an error back to the initiating SIP Server.




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   When the SIP session (and RTP flow) is established, the SIP servers
   initiate accounting records, which are transfered to the AAA
   infrastructure. The accounting records should include usage
   information, that is necessary for billing purposes. The traditional
   telco world current makes use, and prefers, non-cumulative accounting
   information, therefore any interim accounting information MUST be
   non-cumulative. At the end of the SIP session, a final accounting
   record should be issued that includes a summary of the session.

2.2.2  Invitation originating from the mobile node

                  Home Network
           +----------+  +-------+  +----------+
           | xyz.com  |  |xyz.com|  |  Naming  |
             Location |  |  AAAH |  |  Server  |
           | Database |  | server|  | (e.g.DNS)|
           +----------+  +-------+  +----------+
               ^              ^        ^
               |Location      |AAA     |abc.com?
               |Query         |        |
               v              v        v
       +-----------+Invite  +-----------+Invite +-----------+
       |    SIP    +------->|   Anchor  +------>|  abc.com  |
       |  Central  |        |Softswitch/|       |           |
       |  Point of |<-------+    SIP    |<------+    SIP    |
       |  Contact  |200 OK  |   Proxy   |200 OK |   Server  |
       +---+-------+        +-----------+       +------+----+
           |   ^                                    ^  |
           |   |                              200 OK|  |SIP
     200 OK|   |SIP                                 |  |Invite
           |   |Invite                              |  |
           v   |                                    |  v
        +------+--+                               +-+-------+
        |   SIP   |                               |   SIP   |
        | Client  |                               | Client  |
        +---------+                               +---------+
         bob@xyz.com                              joe@abc.com

          Figure 5: Mobile node initiated SIP Session Initiation

   Figure 5 provides an example of a user, bob@xyz.com, that wishes to
   establish a SIP session with joe@abc.com.

   The SIP Client of bob@xyz.com sends the SIP Invite, either to his
   SCPC SIP proxy, or directly to the Anchor SIP proxy, if this is
   known. If the SIP Central Point of Contact receives the SIP Invite,
   it will after a location lookup proxy it to the Anchor SIP proxy.




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   Upon receipt of the SIP Invite message, the Anchor SIP Server MAY
   send an authorization request to the AAAH, in order to determine
   whether the session can be established. The authorization check by
   the AAAH MAY include other policy decisions, such as time of day,
   origination point of the call, etc. A successful response from the
   AAAH will result in the forwarding of the SIP Invite message to the
   SIP Client. A response that includes an error would cause the Anchor
   SIP Server to return an error back to the initiating SIP Client.

   The Anchor SIP Server will after this use ordinary SIP proxying
   mechanisms to proxy the SIP Invite to the SIP server serving
   joe@abc.com, which will proxy the message to the SIP Client of
   joe@abc.com. SIP 200 OK messages traverse back along the same path.

   When the SIP session (and RTP flow) is established, the SIP servers
   initiate accounting records, which are transfered to the AAA
   infrastructure. The accounting records should include usage
   information, that is necessary for billing purposes. The traditional
   telco world current makes use, and prefers, non-cumulative accounting
   information, therefore any interim accounting information MUST be
   non-cumulative. At the end of the SIP session, a final accounting
   record should be issued that includes a summary of the session.


3.0  Requirements

   From the previous section, we can extract the following requirements:

       - A basic requirement for Service Providers is to integrate
         different networks for more efficient operations. IETF AAA
         efforts support this idea by striving for a single AAA
         architecture and protocol set. Whether access is supported by
         PPP, Mobile-IP, MEGACO or SIP, a common architecture and
         protocol set SHOULD be used.
       - The AAA infrastructure MUST be able to perform policy control
         of the SIP servers/proxies, controlling their
         behavior/functionality based on user and/or network policies.
       - The AAAH MUST authenticate a user, and MAY authorize a user and
         the session being requested. The Home AAA server may implement
         whatever policy it wishes on authorizing the call, such as time
         of day, long distance, etc.
       - The AAA infrastructure MUST be able to distribute (push or
         pull) subscriber/service profiles to the relevant SIP servers,
         based on policies.
       - The AAA infrastructure MUST be able to do an allocation of a
         SIP server for a subscriber at registration time, based on
         policies, load distribution etc.
       - Allow SIP messages to be sent through the AAA infrastructure as



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         opaque data (e.g. when the authentication procedures includes
         http digest calculation), to allow the Home AAA server to
         authenticate the message. This eliminates the need for the Home
         AAA server to send the user's "password" to SIP servers.
       - The AAA infrastructure MUST support SIM and smart cards.
       - Ability for SIP Servers to provide the duration of a call, the
         parties involved, and other relevant information to the visited
         and home AAA servers as accounting information.
       - The AAA protocol MUST support for both cumulative, and non-
         cumulative, accounting models.
       - The AAA accounting messages MUST Separate the "session duration
         time" information from those generated via additional services
         (e.g. 3-way calling, etc.)
       - Support for real-time and non-real time accounting data
         transfers.


4.0  Security Considerations

   It is assumed that integrating AAA and IP Telephony/Multimedia will
   not introduce any new security risks. Therefore, the AAA data MUST be
   secured and obscured when transiting the network, the endpoints MUST
   be authenticated before data is sent, and the endpoints MAY be
   authorized to access certain types of AAA data.


5.0  References


[1]  M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg, "SIP: Ses-
     sion Initiation Protocol". RFC 2543, March 1999.

[2]  Aboba et al, "Network Access AAA Evaluation Criteria", IETF work in
     progress, draft-ietf-aaa-na-reqts-02.txt, March 2000.

[3]  S. Bradner, "Key words for use in RFCs to Indicate Requirement Lev-
     els", BCP 14, RFC 2119, March 1997.

[4]  S. Glass, S. Jacobs, C. Perkins, "Mobile IP Authentication, Autho-
     rization, and Accounting Requirements", draft-ietf-mobileip-aaa-
     reqs-03.txt, IETF work in progress, March 2000.

[5]  M. Beadles, D. Mitton, "Criteria for Evaluating Network Access
     Server Protocols", draft-ietf-nasreq-criteria-05.txt, IETF work in
     progress, June 2000.

[6]  E. Guttman, C. Perkins, J. Veizades, M. Day. "Service Location Pro-
     tocol, Version 2", RFC 2165, June 1999.



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[7]  Aboba, Beadles "The Network Access Identifier." RFC 2486.  January
     1999.

[8]  H. Schulzrinne, L. Slutsman, I. Faynberg, H. Lu, "Interworking
     between SIP and INAP". draft-schulzrinne-sin-00.txt, IETF Work in
     Progress, July 2000.

[9]  D. Harkins, D. Carrell, "The Internet Key Exchange (IKE)" RFC 1409,
     November 1998.


6.0  Acknowledgements

   Many of the requirements, and thoughts expressed in this Internet
   Draft, came from presentations that were presented at various 3rd
   Generation wireless meetings, such as 3GPP, 3GPP2 and MWIF.


7.0  Authors' Addresses

   Questions about this memo can be directed to:

      Henrik Basilier
      Ericsson Inc.
      2100 Shattuck Ave.
      Berkeley, Califonia, 94704
      USA

       Phone:  +1 858-361-4314
         Fax:  +1 510-666-3999
      E-mail:  henrik.basilier@ericsson.com


      Pat R. Calhoun
      Network and Security Research Center, Sun Laboratories
      Sun Microsystems, Inc.
      15 Network Circle
      Menlo Park, California, 94025
      USA

       Phone:  +1 650-786-7733
         Fax:  +1 650-786-6445
      E-mail:  pcalhoun@eng.sun.com


      Matt Holdrege
      ipVerse
      223 Ximeno Ave.



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      Long Beach, CA 90803
      U.S.A.

      E-mail:  matt@ipverse.com


      Tony Johansson
      Ericsson Inc.
      2100 Shattuck Ave.
      Berkeley, Califonia, 94704
      USA

       Phone:  +1 510-305-6108
         Fax:  +1 510-666-3999
      E-mail:  tony.johansson@ericsson.com


      James Kempf
      Network and Security Research Center, Sun Laboratories
      Sun Microsystems, Inc.
      15 Network Circle
      Menlo Park, California, 94025
      USA

       Phone:  +1 650-786-5890
         Fax:  +1 650-786-6445
      E-mail:  james.kempf@eng.sun.com



8.0  Full Copyright Statement

   Copyright (C) The Internet Society (2000).  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 docu-
   ment 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 develop-
   ing 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 lim-
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INTERNET DRAFT                                             November 2000


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