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Versions: (draft-saintandre-sip-xmpp-core) 00 01 02 03 04 05 06 07 08 09 10 11 RFC 7247

Network Working Group                                     P. Saint-Andre
Internet-Draft                                       Cisco Systems, Inc.
Intended status: Standards Track                                A. Houri
Expires: June 22, 2014                                               IBM
                                                           J. Hildebrand
                                                     Cisco Systems, Inc.
                                                       December 19, 2013


   Interworking between the Session Initiation Protocol (SIP) and the
    Extensible Messaging and Presence Protocol (XMPP): Architecture,
                     Addresses, and Error Handling
                        draft-ietf-stox-core-09

Abstract

   As a foundation for the definition of bidirectional protocol mappings
   between the Session Initiation Protocol (SIP) and the Extensible
   Messaging and Presence Protocol (XMPP), this document specifies the
   architectural assumptions underlying such mappings as well as the
   mapping of addresses and error conditions.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   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."

   This Internet-Draft will expire on June 22, 2014.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents



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   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Intended Audience  . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   4.  Architectural Assumptions  . . . . . . . . . . . . . . . . . .  4
   5.  Interdomain Federation . . . . . . . . . . . . . . . . . . . .  5
   6.  Address Mapping  . . . . . . . . . . . . . . . . . . . . . . .  6
     6.1.  Overview . . . . . . . . . . . . . . . . . . . . . . . . .  6
     6.2.  Local Part Mapping . . . . . . . . . . . . . . . . . . . .  7
     6.3.  Instance-Specific Mapping  . . . . . . . . . . . . . . . .  9
     6.4.  SIP to XMPP  . . . . . . . . . . . . . . . . . . . . . . .  9
     6.5.  XMPP to SIP  . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  Error Mapping  . . . . . . . . . . . . . . . . . . . . . . . . 11
     7.1.  XMPP to SIP  . . . . . . . . . . . . . . . . . . . . . . . 13
     7.2.  SIP to XMPP  . . . . . . . . . . . . . . . . . . . . . . . 14
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 16
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 16
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 17
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 17
     10.2. Informative References . . . . . . . . . . . . . . . . . . 18
   Appendix A.  Acknowledgements  . . . . . . . . . . . . . . . . . . 19
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 19





















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1.  Introduction

   The IETF has worked on two signalling technologies that can be used
   for multimedia session negotiation, messaging, presence, capabilities
   discovery, notifications, and other application-level functionality:

   o  The Session Initiation Protocol [RFC3261], along with various SIP
      extensions developed within the SIP for Instant Messaging and
      Presence Leveraging Extensions (SIMPLE) Working Group.
   o  The Extensible Messaging and Presence Protocol [RFC6120], along
      with various XMPP extensions developed by the IETF as well as by
      the XMPP Standards Foundation (XSF).

   Because these technologies are widely deployed, it is important to
   clearly define mappings between them for the sake of interworking.
   This document provides the basis for a series of SIP-XMPP
   interworking specifications by defining architectural assumptions,
   address translation methods, and error condition mappings.  Other
   documents in this series define mappings for presence, messaging, and
   media sessions.

   The guidelines in this series are based on implementation and
   deployment experience, and describe techniques that have worked well
   in the field with existing systems.  In practice, interworking has
   been achieved through direct protocol mappings, not through mapping
   to an abstract model as described in, for example, [RFC3859] and
   [RFC3860].  Therefore this series describes the direct mapping
   approach in enough detail for developers of new implementations to
   achieve practical interworking between SIP systems and XMPP systems.


2.  Intended Audience

   The documents in this series are intended for use by software
   developers who have an existing system based on one of these
   technologies (e.g., SIP), and would like to enable communication from
   that existing system to systems based on the other technology (e.g.,
   XMPP).  With regard to this document we assume that readers are
   familiar with the core specifications for both SIP and XMPP, and with
   regard to the other documents in this series we assume that readers
   are familiar with the relevant SIP and XMPP extensions.


3.  Terminology

   A number of terms used here are explained in [RFC3261] and [RFC6120].

   Several examples use the "XML Notation" from the IRI specification



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   [RFC3987] to represent Unicode characters outside the ASCII range
   (e.g., the string "ue" stands for the Unicode character LATIN SMALL
   LETTER U WITH DIAERESIS, U+00FC).

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in
   [RFC2119].


4.  Architectural Assumptions

   Protocol translation between SIP and XMPP could occur in a number of
   different entities, depending on the architecture of real-time
   communication deployments.  For example, protocol translation could
   occur within a multi-protocol server (which uses protocol-specific
   connection managers to initiate traffic to and accept traffic from
   clients or other servers natively using SIP/SIMPLE, XMPP, etc.),
   within a multi-protocol client (which enables a user to establish
   connections natively with various servers using SIP/SIMPLE, XMPP,
   etc.), or within a gateway that acts as a dedicated protocol
   translator (which takes one protocol as input and provides another
   protocol as output).

   This document assumes that the protocol translation will occur within
   a gateway, specifically:

   o  When information needs to flow from an XMPP-based system to a SIP-
      based system, protocol translation will occur within an "XMPP-to-
      SIP gateway" that translates XMPP syntax and semantics on behalf
      of an "XMPP server" (together, these two logical functions
      comprise an "XMPP service").
   o  When information needs to flow from a SIP-based system to an XMP-
      based system, protocol translation will occur within a "SIP-to-
      XMPP gateway" that translates SIP syntax and semantics on behalf
      of a "SIP server" (together, these two logical functions comprise
      a "SIP service").

   Naturally, these logical functions could occur in one and the same
   actual entity; we differentiate between them mainly for explanatory
   purposes (although, in practice, such gateways are indeed fairly
   common).

      Note: This assumption is not meant to discourage protocol
      translation within multi-protocol clients or servers; instead,
      this assumption is followed mainly to clarify the discussion and
      examples so that the protocol translation principles can be more
      easily understood and can be applied by client and server



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      implementors with appropriate modifications to the examples and
      terminology.

   This document assumes that a gateway will translate directly from one
   protocol to the other.  For the sake of the examples, we further
   assume that protocol translation will occur within a gateway in the
   source domain, so that information generated by the user of an XMPP
   system will be translated by a gateway within the trust domain of
   that XMPP system, and information generated by the user of a SIP
   system will be translated by a gateway within the trust domain of
   that SIP system.  However, nothing in this document ought to be taken
   as recommending against protocol translation at the destination
   domain.

   An architectural diagram for a possible gateway deployment is shown
   below, where the entities have the following significance and the "#"
   character is used to show the boundary of a trust domain:

   o  romeo@example.net -- a SIP user.
   o  example.net -- a SIP server with an associated gateway ("S2X GW")
      to XMPP.
   o  juliet@example.com -- an XMPP user.
   o  example.com -- an XMPP server with an associated gateway ("X2S
      GW") to SIP.

      ##@######################################################
      #                           :                           #
      #                 +-----+   :                           #
      #                 | S2X |   :                           #
      #   +-------------+ GW  |---:-------->+-------------+   #
      #   | SIP Server  +-----+   :         | XMPP Server |   #
      #   | example.net |         :   +-----+ example.com |   #
      #   +-------------+<--------:---| X2S +-------------+   #
      #         |                 :   | GW  |  |              #
      #         |                 :   +-----+  |              #
      #         |                 :            |              #
      #    romeo@example.net      :      juliet@example.com   #
      #                           :                           #
      #########################################################


5.  Interdomain Federation

   The architecture assumptions underlying this document imply that
   communication between a SIP system and an XMPP system will take place
   using interdomain federation: the SIP server invokes its associated
   SIP-to-XMPP gateway, which communicates with the XMPP server using
   native XMPP server-to-server methods; similarly, the XMPP server



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   invokes its associated XMPP-to-SIP gateway, which communicates with
   the SIP server using native SIP server-to-server methods.

   When an XMPP server receives an XMPP stanza whose 'to' address
   specifies or includes a domain other than the domain of the XMPP
   server, it needs to determine whether the destination domain
   communicates via XMPP or SIP.  To do so, it performs one or more DNS
   SRV lookups [RFC2782] for "_xmpp-server" records as specified in
   [RFC6120].  If the response returns a hostname, the XMPP server can
   attempt XMPP communication.  If not, it can determine the appropriate
   location for SIP communication at the target domain using the
   procedures specified in [RFC3263].

   Similarly, when a SIP server receives a SIP message whose Request-URI
   specifies or includes a domain other than the domain of the SIP
   server, it needs to determine whether the destination domain
   communicates via SIP or XMPP.  To do so, it uses the procedures
   specified in [RFC3263].  If that response returns a hostname, the SIP
   server can attempt SIP communication.  If not, it can perform one or
   more DNS SRV lookups [RFC2782] for "_xmpp-server" records as
   specified in [RFC6120].

   In both cases, the server in question might have previously
   determined that the foreign domain communicates via SIP or XMPP, in
   which case it would not need to perform the relevant DNS lookups.
   The caching of such information is a matter of implementation and
   local service policy, and is therefore out of scope for this
   document.

   Because [RFC6120] specifies a binding of XMPP to TCP, a gateway from
   SIP to XMPP will need to support TCP as the underlying transport
   protocol.  By contrast, as specified in [RFC3261], either TCP or UDP
   can be used as the underlying transport for SIP messages, and a given
   SIP deployment might support only UDP; therefore, a gateway from XMPP
   to SIP might need to communicate with a SIP server using either TCP
   or UDP.


6.  Address Mapping

6.1.  Overview

   The basic SIP address format is a 'sip' or 'sips' URI as specified in
   [RFC3261].  When a SIP entity supports extensions for instant
   messaging it might be identified by an 'im' URI as specified in the
   Common Profile for Instant Messaging [RFC3860] (see [RFC3428]) and
   when a SIP entity supports extensions for presence it might be
   identified by a 'pres' URI as specified in the Common Profile for



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   Presence [RFC3859] (see [RFC3856]).  SIP entities typically also
   support the 'tel' URI scheme [RFC3966] and might support other URI
   schemes as well.

   The XMPP address format is specified in [RFC6122] (although note that
   XMPP URIs [RFC5122] are not used natively on the XMPP network); in
   addition, [RFC6121] encourages instant messaging and presence
   applications of XMPP to also support 'im' and 'pres' URIs as
   specified in [RFC3860] and [RFC3859] respectively, although such
   support might simply involve leaving resolution of such addresses up
   to an XMPP server.

   In this document we primarily describe mappings for addresses of the
   form <user@domain>; however, we also provide guidelines for mapping
   the addresses of specific user agent instances, which take the form
   of Globally Routable User Agent URIs (GRUUs) in SIP and
   "resourceparts" in XMPP.  Mapping of protocol-specific identifiers
   (such as telephone numbers) is out of scope for this specification.
   In addition, we have ruled the mapping of domain names as out of
   scope for now since that is a matter for the Domain Name System;
   specifically, the issue for interworking between SIP and XMPP relates
   to the translation of fully internationalized domain names (IDNs)
   into non-internationalized domain names (IDNs are not allowed in the
   SIP address format, but are allowed in the XMPP address via
   Internationalized Domain Names in Applications, see [RFC6122] and
   [I-D.ietf-xmpp-6122bis]).  Therefore, in the following sections we
   focus primarily on the local part of an address (these are called
   variously "usernames", "instant inboxes", "presentities", and
   "localparts" in the protocols at issue), secondarily on the instance-
   specific part of an address, and not at all on the domain-name part
   of an address.

   The sip:/sips:, im:/pres:, and XMPP address schemes allow different
   sets of characters (although all three allow alphanumeric characters
   and disallow both spaces and control characters).  In some cases,
   characters allowed in one scheme are disallowed in others; these
   characters need to be mapped appropriately in order to ensure
   interworking across systems.

6.2.  Local Part Mapping

   The local part of a sip:/sips: URI inherits from the "userinfo" rule
   in [RFC3986] with several changes; here we discuss the SIP "user"
   rule only:







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      user             =  1*( unreserved / escaped / user-unreserved )
      user-unreserved  =  "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"
      unreserved       =  alphanum / mark
      mark             =  "-" / "_" / "." / "!" / "~" / "*" / "'"
                          / "(" / ")"

   Here we make the simplifying assumption that the local part of an
   im:/pres: URI inherits from the "dot-atom-text" rule in [RFC5322]
   rather than the more complicated "local-part" rule:

      dot-atom-text =  1*atext *("." 1*atext)
      atext         =  ALPHA / DIGIT /    ; Any character except
                       "!" / "#" / "$" /  ; controls, SP, and
                       "%" / "&" / "'" /  ; specials. Used for
                       "*" / "+" / "-" /  ; atoms.
                       "/" / "=" / "?" /
                       "^" / "_" / "`" /
                       "{" / "|" / "}" /
                       "~"

   The local part of an XMPP address allows any ASCII character except
   space, controls, and the " & ' / : < > @ characters.

   To summarize the foregoing information, the following table lists the
   allowed and disallowed characters in the local part of identifiers
   for each protocol (aside from the alphanumeric, space, and control
   characters), in order by hexadecimal character number (where each "A"
   row shows the allowed characters and each "D" row shows the
   disallowed characters).

   Table 1: Allowed and disallowed characters

   +---+----------------------------------+
   | SIP/SIPS CHARACTERS                  |
   +---+----------------------------------+
   | A | !  $ &'()*+,-./ ; = ?     _    ~ |
   | D |  "# %          : < > @[\]^ `{|}  |
   +---+----------------------------------+
   | IM/PRES CHARACTERS                   |
   +---+----------------------------------+
   | A | ! #$%&'  *+ - /   = ?    ^_`{|}~ |
   | D |  "     ()  , . :;< > @[\]        |
   +---+----------------------------------+
   | XMPP CHARACTERS                      |
   +---+----------------------------------+
   | A | ! #$%  ()*+,-.  ; = ? [\]^_`{|}~ |
   | D |  "   &'       /: < > @           |
   +---+----------------------------------+



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   When transforming the local part of an address from one scheme to
   another, an application SHOULD proceed as follows:

   1.  Unescape any escaped characters in the source address (e.g., from
       SIP to XMPP unescape "%23" to "#" per [RFC3986] and from XMPP to
       SIP unescape "\27" to "'" per [XEP-0106]).
   2.  Leave unmodified any characters that are allowed in the
       destination scheme.
   3.  Escape any characters that are allowed in the source scheme but
       reserved in the destination scheme, as escaping is defined for
       the destination scheme.  In particular:
       *  Where the destination scheme is a URI (i.e., an im:, pres:,
          sip:, or sips: URI), each reserved character MUST be percent-
          encoded to "%hexhex" as specified in Section 2.5 of [RFC3986]
          (e.g., when transforming from XMPP to SIP, encode "#" as
          "%23").
       *  Where the destination scheme is a native XMPP address, each
          reserved character MUST be encoded to "\hexhex" as specified
          in [XEP-0106] (e.g., when transforming from SIP to XMPP,
          encode "'" as "\27").

6.3.  Instance-Specific Mapping

   The meaning of a resourcepart in XMPP (i.e., the portion of a JID
   after the slash character, such as "foo" in "user@example.com/foo")
   matches that of a Globally Routable User Agent URI (GRUU) in SIP
   [RFC5627].  In both cases, these constructs identify a particular
   device associated with the bare JID ("localpart@domainpart") of an
   XMPP entity or with the Address of Record (AOR) of a SIP entity.
   Therefore, it is reasonable to map the value of a "gr" URI parameter
   to an XMPP resourcepart, and vice-versa.

   The mapping described here does not apply to temporary GRUUs, only to
   GRUUs associated with an Address of Record.

   The "gr" URI parameter in SIP can contain only characters from the
   ASCII range (although characters outside the ASCII range can be
   percent-encoded in accordance with [RFC3986]), whereas an XMPP
   resourcepart can contain nearly any Unicode character [UNICODE].
   Therefore Unicode characters outside the ASCII range need to be
   mapped to characters in the ASCII range, as described below.

6.4.  SIP to XMPP

   The following is a high-level algorithm for mapping a sip:, sips:,
   im:, or pres: URI to an XMPP address:





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   1.  Remove URI scheme.
   2.  Split at the first '@' character into local part and hostname
       (mapping the latter is out of scope).
   3.  Translate any percent-encoded strings ("%hexhex") to percent-
       decoded octets.
   4.  Treat result as a UTF-8 string.
   5.  Translate "&" to "\26", "'" to "\27", and "/" to "\2f"
       respectively in order to properly handle the characters
       disallowed in XMPP addresses but allowed in sip:/sips: URIs and
       im:/pres: URIs as shown in Table 1 above (this is consistent with
       [XEP-0106]).
   6.  Apply Nodeprep profile of Stringprep [RFC3454] or its replacement
       (see [RFC6122] and [I-D.ietf-xmpp-6122bis]) for canonicalization
       (OPTIONAL).
   7.  Recombine local part with mapped hostname to form a bare JID
       ("localpart@domainpart").
   8.  If the (SIP) address contained a "gr" URI parameter, append a
       slash character "/" and the "gr" value to the bare JID to form a
       full JID ("localpart@domainpart/resourcepart").

   Several examples follow, illustrating steps 3, 5, and 8 described
   above.

      +----------------------------+--------------------------+
      | SIP URI                    |  XMPP Address            |
      +----------------------------+--------------------------+
      | sip:f%C3%BC@sip.example    |  f&#xFC;@sip.example     |
      | sip:o'malley@sip.example   |  o\27malley@sip.example  |
      | sip:foo@sip.example;gr=bar |  foo@sip.example/bar     |
      +----------------------------+--------------------------+

   In the first example the string "%C3%BC" is a percent-encoded
   representation of the UTF-8-encoded Unicode character LATIN SMALL
   LETTER U WITH DIAERESIS (U+00FC), whereas the string "&#xFC;" is the
   same character shown for documentation purposes using the XML
   Notation defined in [RFC3987] (in XMPP it would be sent directly as a
   UTF-8-encoded Unicode character and not percent-encoded as in a SIP
   URI to comply with the URI syntax defined in [RFC3986]).

6.5.  XMPP to SIP

   The following is a high-level algorithm for mapping an XMPP address
   to a sip:, sips:, im:, or pres: URI:

   1.  Split XMPP address into localpart (mapping described in remaining
       steps), domainpart (hostname; mapping is out of scope), and
       resourcepart (specifier for particular device or connection, for
       which an OPTIONAL mapping is described below).



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   2.  Apply Nodeprep profile of [RFC3454] or its replacement (see
       [RFC6122] and [I-D.ietf-xmpp-6122bis]) for canonicalization of
       the XMPP localpart (OPTIONAL).
   3.  Translate "\26" to "&", "\27" to "'", and "\2f" to "/"
       respectively (this is consistent with [XEP-0106]).
   4.  Determine if the foreign domain supports im: and pres: URIs
       (discovered via [RFC2782] lookup as specified in [RFC6121]), else
       assume that the foreign domain supports sip:/sips: URIs.
   5.  If converting into im: or pres: URI, for each byte, if the byte
       is in the set (),.;[\] or is a UTF-8 character outside the ASCII
       range then percent-encode that byte to "%hexhex" format.  If
       converting into sip: or sips: URI, for each byte, if the byte is
       in the set #%[\]^`{|} or is a UTF-8 character outside the ASCII
       range then percent-encode that byte to "%hexhex" format.
   6.  Combine resulting local part with mapped hostname to form
       local@domain address.
   7.  Prepend with 'im:' scheme (for XMPP <message/> stanzas) or
       'pres:' scheme (for XMPP <presence/> stanzas) if foreign domain
       supports these, else prepend with 'sip:' or 'sips:' scheme
       according to local service policy.
   8.  If the XMPP address included a resourcepart and the destination
       URI scheme is 'sip:' or 'sips:', optionally append the slash
       character '/' and then append the resourcepart (making sure to
       percent-encode any UTF-8 characters outside the ASCII range) as
       the "gr" URI parameter.

   Several examples follow, illustrating steps 3, 5, and 8 described
   above.

      +---------------------------+---------------------------------+
      | XMPP Address              |  SIP URI                        |
      +---------------------------+---------------------------------+
      | m\26m@xmpp.example        |  sip:m&m@xmpp.example           |
      | tsch&#xFC;ss@xmpp.example |  sip:tsch%C3%BCss@xmpp.example  |
      | baz@xmpp.example/qux      |  sip:baz@xmpp.example;gr=qux    |
      +---------------------------+---------------------------------+

   As above, in the first example the string "&#xFC;" is the Unicode
   character LATIN SMALL LETTER U WITH DIAERESIS (U+00FC) shown for
   documentation purposes using the XML Notation defined in [RFC3987]
   (in XMPP it would be sent directly as a UTF-8-encoded Unicode
   character and not percent-encoded, whereas the string "%C3%BC" is a
   percent-encoded representation of the of the same character.


7.  Error Mapping

   Various differences between XMPP error conditions and SIP response



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   codes make it hard to provide a comprehensive and consistent mapping
   between the protocols:

   o  Whereas the set of XMPP error conditions is fixed in the core XMPP
      specification (and supplemented where needed by application-
      specific extensions), the set of SIP response codes is more open
      to change, as evidenced by the IANA registry of SIP response
      codes.
   o  XMPP has defined fewer error conditions related to stanza handling
      (22 are defined in [RFC6120]) than SIP has defined response codes
      related to message handling (at the date of this writing, 71 SIP
      response codes are registered with IANA as defined in [RFC3261]
      and numerous SIP extensions).
   o  In many cases, the SIP response codes are more specific than the
      XMPP error conditions (e.g., from an XMPP perspective the SIP
      codes "413 Request Entity Too Large" and "414 Request-URI Too
      Long" are just two forms of a bad request, and the SIP codes "415
      Unsupported Media Type" and "416 Unsupported URI Scheme" are just
      two forms of a request that is not acceptable).
   o  SIP differentiates between responses about a particular endpoint
      or resource (the 4xx series) and responses about a user, i.e., all
      of a user's endpoints or resources (the 6xx series).  There is no
      such distinction in XMPP, since the same error condition can be
      returned in relation to the "bare JID" (localpart@domainpart) of a
      user or the "full JID" (localpart@domainpart/resourcepart) of a
      particular endpoint or resource, depending on the 'to' address of
      the original request.

   As a result of these and other factors, the mapping of error
   conditions and response codes is more of an art than a science.  This
   document provides suggested mappings, but implementations are free to
   deviate from these mappings if needed.  Also, because no XMPP error
   conditions are equivalent to the provisional (1xx) and successful
   (2xx) response codes in SIP, this document suggests mappings only for
   the SIP redirection (3xx), request failure (4xx), server failure
   (5xx), and global failure (6xx) response code families.

   Supplementary information about SIP response codes can be expressed
   in the "Reason-Phrase" in the Status-Line header, and detailed
   information about XMPP error conditions can be expressed in the
   <text/> child of the <error/> element.  Although the semantics of
   these constructs are specified in a slightly different way, it is
   reasonable for a gateway to map these constructs to each other if
   they are found in a SIP response or XMPP error stanza.







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7.1.  XMPP to SIP

   The mapping of specific XMPP error conditions to SIP response codes
   SHOULD be as described in the following table.

   Table 2: Mapping of XMPP error conditions to SIP response codes

      +------------------------------+---------------------+
      |  XMPP Error Condition        |  SIP Response Code  |
      +------------------------------+---------------------+
      |  <bad-request/>              | 400                 |
      |  <conflict/>                 | 400                 |
      |  <feature-not-implemented/>  | 405 or 501 (1)      |
      |  <forbidden/>                | 403 or 603 (2)      |
      |  <gone/>                     | 301 or 410 (3)      |
      |  <internal-server-error/>    | 500                 |
      |  <item-not-found/>           | 404 or 604 (2)      |
      |  <jid-malformed/>            | 400                 |
      |  <not-acceptable/>           | 406 or 606 (2)      |
      |  <not-allowed/>              | 403                 |
      |  <not-authorized/>           | 401                 |
      |  <policy-violation/>         | 403                 |
      |  <recipient-unavailable/>    | 480 or 600 (2)      |
      |  <redirect/>                 | 302                 |
      |  <registration-required/>    | 407                 |
      |  <remote-server-not-found/>  | 404 or 408 (4)      |
      |  <remote-server-timeout/>    | 408                 |
      |  <resource-constraint/>      | 500                 |
      |  <service-unavailable/>      | see note (5) below  |
      |  <subscription-required/>    | 400                 |
      |  <undefined-condition/>      | 400                 |
      |  <unexpected-request/>       | 491 or 400          |
      +------------------------------+---------------------+

   1.  If the error relates to a "full JID"
       (localpart@domainpart/resourcepart), the SIP 405 response code is
       RECOMMENDED.  If the error relates to a "bare JID"
       (localpart@domainpart), the SIP 501 response code is RECOMMENDED.
   2.  If the error relates to a "full JID"
       (localpart@domainpart/resourcepart), the SIP response code from
       the 4xx series is RECOMMENDED.  If the error relates to a "bare
       JID" (localpart@domainpart), the SIP response code from the 6xx
       series is RECOMMENDED.
   3.  If the <gone/> element includes XML character data specifying the
       new address, the error MUST be mapped to SIP 301; if not, it MUST
       be mapped to SIP 410.





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   4.  The XMPP <remote-server-not-found/> error can mean either that
       the remote server (a) does not exist or (b) cannot be resolved.
       SIP has two different response codes here, 404 to cover (a) and
       408 to cover (b).
   5.  The XMPP <service-unavailable/> error condition is widely used to
       inform the requesting entity that the intended recipient does not
       support the relevant feature, to signal that a server cannot
       perform the requested service either generally or in relation to
       a particular user, and to avoid disclosing whether a given
       account exists at all.  This is quite different from the
       semantics of the SIP 503 Service Unavailable response code, which
       is used to signal that communication with a server is impossible
       (e.g., even if the XMPP <service-unavailable/> error condition is
       returned in relation to a specific user, the SIP 503 response
       code will be interpreted as applying to all future requests to
       this server, not just requests for the specific user).
       Therefore, mapping the XMPP <service-unavailable/> error
       condition to the SIP 503 Service Unavailable response code is NOT
       RECOMMENDED.  Although no precise mapping is available, the SIP
       403 Forbidden and 405 Method Not Allowed response codes are
       closest in meaning to the XMPP <service-unavailable/> error
       condition.

7.2.  SIP to XMPP

   The mapping of SIP response codes to XMPP error conditions SHOULD be
   as described in the following table.  If a gateway encounters a SIP
   response code that is not listed below, it SHOULD map a 3xx-series
   code to <redirect/>, a 4xx-series code to <bad-request/>, a 5xx-
   series code to <internal-server-error>, and a 6xx-series code to
   <recipient-unavailable/>.

   Table 3: Mapping of SIP response codes to XMPP error conditions

      +---------------------+---------------------------------+
      |  SIP Response Code  |  XMPP Error Condition           |
      +---------------------+---------------------------------+
      |  3xx                |  <redirect/>                    |
      |  300                |  <redirect/>                    |
      |  301                |  <gone/> (1)                    |
      |  302                |  <redirect/>                    |
      |  305                |  <redirect/>                    |
      |  380                |  <not-acceptable/>              |
      |  4xx                |  <bad-request/>                 |
      |  400                |  <bad-request/>                 |
      |  401                |  <not-authorized/>              |
      |  402                |  <bad-request/> (2)             |
      |  403                |  <forbidden/> (3)               |



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      |  404                |  <item-not-found/> (4)          |
      |  405                |  <feature-not-implemented/>     |
      |  406                |  <not-acceptable/>              |
      |  407                |  <registration-required/>       |
      |  408                |  <remote-server-timeout/> (5)   |
      |  410                |  <gone/> (1)                    |
      |  413                |  <policy-violation/>            |
      |  414                |  <policy-violation/>            |
      |  415                |  <not-acceptable/>              |
      |  416                |  <not-acceptable/>              |
      |  420                |  <feature-not-implemented/>     |
      |  421                |  <not-acceptable/>              |
      |  423                |  <resource-constraint/>         |
      |  430                |  <recipient-unavailable/> (6)   |
      |  439                |  <feature-not-implemented/> (6) |
      |  440                |  <policy-violation/> (7)        |
      |  480                |  <recipient-unavailable/>       |
      |  481                |  <item-not-found/>              |
      |  482                |  <not-acceptable/>              |
      |  483                |  <not-acceptable/>              |
      |  484                |  <item-not-found/>              |
      |  485                |  <item-not-found/>              |
      |  486                |  <recipient-unavailable/>       |
      |  487                |  <recipient-unavailable/>       |
      |  488                |  <not-acceptable/>              |
      |  489                |  <policy-violation/> (8)        |
      |  491                |  <unexpected-request/>          |
      |  493                |  <bad-request/>                 |
      |  5xx                |  <internal-server-error/>       |
      |  500                |  <internal-server-error/>       |
      |  501                |  <feature-not-implemented/>     |
      |  502                |  <remote-server-not-found/>     |
      |  503                |  <internal-server-error/> (9)   |
      |  504                |  <remote-server-timeout/>       |
      |  505                |  <not-acceptable/>              |
      |  513                |  <policy-violation/>            |
      |  6xx                |  <recipient-unavailable/>       |
      |  600                |  <recipient-unavailable/>       |
      |  603                |  <recipient-unavailable/>       |
      |  604                |  <item-not-found/>              |
      |  606                |  <not-acceptable/>              |
      +---------------------+---------------------------------+

   1.  When mapping SIP 310 to XMPP <gone/>, the <gone/> element MUST
       include XML character data specifying the new address.  When
       mapping SIP 410 to XMPP <gone/>, the <gone/> element MUST NOT
       include XML character data specifying a new address.




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   2.  The XMPP <payment-required/> error condition was removed in
       [RFC6120].  Therefore, a mapping to XMPP <bad-request/>.
   3.  Depending on the scenario, other possible translations for SIP
       403 are <not-allowed/> and <policy-violation/>.
   4.  Depending on the scenario, another possible translation for SIP
       404 is <remote-sever-not-found/>.
   5.  Depending on the scenario, another possible translation for SIP
       408 is <remote-server-not-found/>.
   6.  Codes 430 and 439 are defined in [RFC5626].
   7.  Code 440 is defined in [RFC5393].
   8.  Code 489 is defined in [RFC6665].
   9.  Regarding the semantic mismatch between XMPP <service-
       unavailable/> and SIP code 503, see note under Section 6.1 of
       this document.


8.  IANA Considerations

   This document makes no requests of IANA.


9.  Security Considerations

   Detailed security considerations for SIP are given in [RFC3261] and
   for XMPP in [RFC6120].

   As specified in Section 26.4.4 of [RFC3261] and updated by [RFC5630],
   a To header or a Request-URI containing a SIPS URI is used to
   indicate that all hops in a communication path need to be protected
   using Transport Layer Security [RFC5246].  Because XMPP lacks a way
   to signal that all hops need to be encrypted, if the To header or
   Request-URI of a SIP message is a SIPS URI then the SIP-to-XMPP
   gateway MUST NOT translate the SIP message into an XMPP stanza and
   MUST NOT route it to the destination XMPP server.

   A gateway between SIP and XMPP (in either direction) effectively acts
   as a SIP back-to-back user agent ("B2BUA").  The amplification
   vulnerability described in [RFC5393] can manifest itself with B2BUAs
   (see also [I-D.ietf-straw-b2bua-loop-detection]), and a gateway
   SHOULD implement the loop-detection methods defined in that
   specification to help mitigate the possibility of amplification
   attacks.  Note that, although it would be possible to signal the Max-
   Forwards and Max-Breadth SIP headers over XMPP using the Stanza
   Headers and Internet Metadata (SHIM) extension [XEP-0131], that
   extension is not widely implemented; therefore, defenses against
   excessive looping and amplification attacks when messages pass back
   and forth through SIP and XMPP networks is out of scope for this
   document.  However, it ought to be addressed in the future, and



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   implementations are strongly encouraged to incorporate appropriate
   counter measures wherever possible.


10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  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.

   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation
              Protocol (SIP): Locating SIP Servers", RFC 3263,
              June 2002.

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66,
              RFC 3986, January 2005.

   [RFC3987]  Duerst, M. and M. Suignard, "Internationalized Resource
              Identifiers (IRIs)", RFC 3987, January 2005.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

   [RFC5393]  Sparks, R., Lawrence, S., Hawrylyshen, A., and B. Campen,
              "Addressing an Amplification Vulnerability in Session
              Initiation Protocol (SIP) Forking Proxies", RFC 5393,
              December 2008.

   [RFC5627]  Rosenberg, J., "Obtaining and Using Globally Routable User
              Agent URIs (GRUUs) in the Session Initiation Protocol
              (SIP)", RFC 5627, October 2009.

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630, October 2009.

   [RFC6120]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 6120, March 2011.

   [RFC6122]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Address Format", RFC 6122, March 2011.




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   [UNICODE]  The Unicode Consortium, "The Unicode Standard, Version
              6.2", 2012,
              <http://www.unicode.org/versions/Unicode6.2.0/>.

10.2.  Informative References

   [I-D.ietf-straw-b2bua-loop-detection]
              Kaplan, H. and V. Pascual, "Loop Detection Mechanisms for
              Session Initiation Protocol (SIP) Back-to- Back User
              Agents (B2BUAs)", draft-ietf-straw-b2bua-loop-detection-03
              (work in progress), December 2013.

   [RFC2782]  Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
              specifying the location of services (DNS SRV)", RFC 2782,
              February 2000.

   [RFC3428]  Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C.,
              and D. Gurle, "Session Initiation Protocol (SIP) Extension
              for Instant Messaging", RFC 3428, December 2002.

   [RFC3454]  Hoffman, P. and M. Blanchet, "Preparation of
              Internationalized Strings ("STRINGPREP")", RFC 3454,
              December 2002.

   [RFC3856]  Rosenberg, J., "A Presence Event Package for the Session
              Initiation Protocol (SIP)", RFC 3856, August 2004.

   [RFC3859]  Peterson, J., "Common Profile for Presence (CPP)",
              RFC 3859, August 2004.

   [RFC3860]  Peterson, J., "Common Profile for Instant Messaging
              (CPIM)", RFC 3860, August 2004.

   [RFC3966]  Schulzrinne, H., "The tel URI for Telephone Numbers",
              RFC 3966, December 2004.

   [RFC5122]  Saint-Andre, P., "Internationalized Resource Identifiers
              (IRIs) and Uniform Resource Identifiers (URIs) for the
              Extensible Messaging and Presence Protocol (XMPP)",
              RFC 5122, February 2008.

   [RFC5322]  Resnick, P., Ed., "Internet Message Format", RFC 5322,
              October 2008.

   [RFC5626]  Jennings, C., Mahy, R., and F. Audet, "Managing Client-
              Initiated Connections in the Session Initiation Protocol
              (SIP)", RFC 5626, October 2009.




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   [RFC6121]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Instant Messaging and Presence",
              RFC 6121, March 2011.

   [RFC6665]  Roach, A., "SIP-Specific Event Notification", RFC 6665,
              July 2012.

   [I-D.ietf-xmpp-6122bis]
              Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Address Format",
              draft-ietf-xmpp-6122bis-09 (work in progress),
              November 2013.

   [XEP-0106]
              Saint-Andre, P. and J. Hildebrand, "JID Escaping", XSF
              XEP 0106, June 2007.

   [XEP-0131]
              Saint-Andre, P. and J. Hildebrand, "Stanza Headers and
              Internet Metadata", XSF XEP 0131, July 2006.


Appendix A.  Acknowledgements

   The authors wish to thank the following individuals for their
   feedback: Mary Barnes, Dave Cridland, Mike De Vries, Fabio Forno,
   Adrian Georgescu, Philipp Hancke, Saul Ibarra Corretge, Markus
   Isomaki, Olle Johansson, Paul Kyzivat, Salvatore Loreto, Daniel-
   Constantin Mierla, Tory Patnoe, and Robert Sparks.

   Dan Romascanu reviewed the document on behalf of the General Area
   Review Team.

   The authors gratefully acknowledge the assistance of Markus Isomaki
   and Yana Stamcheva as the working group chairs and Gonzalo Camarillo
   as the sponsoring Area Director.















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Authors' Addresses

   Peter Saint-Andre
   Cisco Systems, Inc.
   1899 Wynkoop Street, Suite 600
   Denver, CO  80202
   USA

   Phone: +1-303-308-3282
   Email: psaintan@cisco.com


   Avshalom Houri
   IBM
   Rorberg Building, Pekris 3
   Rehovot  76123
   Israel

   Email: avshalom@il.ibm.com


   Joe Hildebrand
   Cisco Systems, Inc.
   1899 Wynkoop Street, Suite 600
   Denver, CO  80202
   USA

   Email: jhildebr@cisco.com























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