Network Working Group                                            E. Ivov
Internet-Draft                                                     Jitsi
Intended status: Informational                            P. Saint-Andre
Expires: November 03, 2013                           Cisco Systems, Inc.
                                                              E. Marocco
                                                          Telecom Italia
                                                            May 02, 2013

  CUSAX: Combined Use of the Session Initiation Protocol (SIP) and the
           Extensible Messaging and Presence Protocol (XMPP)


   This document describes suggested practices for combined use of the
   Session Initiation Protocol (SIP) and the Extensible Messaging and
   Presence Protocol (XMPP).  Such practices aim to provide a single
   fully featured real-time communication service by using complementary
   subsets of features from each of the protocols.  Typically such
   subsets would include telephony capabilities from SIP and instant
   messaging and presence capabilities from XMPP.  This specification
   does not define any new protocols or syntax for either SIP or XMPP.
   However, implementing it may require modifying or at least
   reconfiguring existing client and server-side software.  Also, it is
   not the purpose of this document to make recommendations as to
   whether or not such combined use should be preferred to the
   mechanisms provided natively by each protocol (for example, SIP's
   SIMPLE or XMPP's Jingle).  It merely aims to provide guidance to
   those who are interested in such a combined use.

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

   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 November 03, 2013.

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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
   ( in effect on the date of
   publication of this document.  Please review these documents
   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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Client Bootstrap  . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Operation . . . . . . . . . . . . . . . . . . . . . . . . . .   6
     3.1.  Server-Side Setup . . . . . . . . . . . . . . . . . . . .   7
     3.2.  Client-Side Discovery and Usability . . . . . . . . . . .   7
     3.3.  Indicating a Relation Between SIP and XMPP Accounts . . .   8
     3.4.  Matching Incoming SIP Calls to XMPP JIDs  . . . . . . . .   9
   4.  Multi-Party Interactions  . . . . . . . . . . . . . . . . . .   9
   5.  Federation  . . . . . . . . . . . . . . . . . . . . . . . . .  10
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  12
   8.  Informative References  . . . . . . . . . . . . . . . . . . .  12
   Appendix A.  Acknowledgements . . . . . . . . . . . . . . . . . .  13
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  14

1.  Introduction

   Historically SIP [RFC3261] and XMPP [RFC6120] have often been
   implemented and deployed with different purposes: from its very start
   SIP's primary goal has been to provide a means of conducting
   "Internet telephone calls".  XMPP on the other hand, has, from its
   Jabber days, been mostly used for instant messaging and presence
   [RFC6121], as well as related services such as groupchat rooms

   For various reasons, these trends have continued through the years
   even after each of the protocols had been equipped to provide the
   features it was initially lacking:

   o  Today, in the context of the SIMPLE working group, the IETF has
      defined a number of protocols and protocol extensions that not

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      only allow for SIP to be used for regular instant messaging and
      presence but that also provide mechanisms for elaborated features
      such as multi-user chats, server-stored contact lists, file
      transfer and others.

   o  Similarly, the XMPP community and the XMPP Standards Foundation
      have worked on defining a number of XMPP Extension Protocols
      (XEPs) that provide XMPP implementations with the means of
      establishing end-to-end sessions.  These extensions are often
      jointly referred to as Jingle and arguably their most popular use
      case is audio and video calling.

   Despite these advances, SIP remains the protocol of choice for
   telephony-like services, especially in enterprises where users are
   accustomed to features such as voice mail, call park, call queues,
   conference bridges and many others that are rarely (if at all)
   available in Jingle-based software.  XMPP implementations, on the
   other hand, greatly outnumber and outperform those available for
   instant messaging and presence extensions developed in the SIMPLE WG,
   such as MSRP [RFC4975] and XCAP [RFC4825].

   For these reasons, in a number of cases adopters have found
   themselves needing a set of features that are not offered by any
   single-protocol solution but that separately exist in SIP and XMPP
   products.  The idea of seamlessly using both protocols together would
   hence often appeal to service providers.  Most often, such a service
   would employ SIP exclusively for audio, video, and telephony services
   and rely on XMPP for anything else varying from chat, contact list
   management, and presence to whiteboarding and exchanging files.
   Because these services and clients involve the combined use of SIP
   and XMPP, we label them "CUSAX" for short.

   +------------+      +-------------+
   | SIP Server |      | XMPP Server |
   +------------+      +-------------+
            \             /
   media     \           /  instant messaging,
   signaling  \         /   presence, etc.
               \       /
            | CUSAX Client |

                  Figure 1: Division of Responsibilities

   This document explains how such hybrid offerings can be achieved with
   a minimum of modifications to existing software while providing an
   optimal user experience.  It covers server discovery, determining a

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   SIP AOR while using XMPP, and determining an XMPP Jabber Identifier
   ("JID") from incoming SIP requests.  Most of the text here pertains
   to client behavior but it also recommends certain server-side

   Note that this document is focused on coexistence of SIP and XMPP
   functionality in end-user-oriented clients.  By intent it does not
   define methods for protocol-level mapping between SIP and XMPP, as
   might be used within a server-side gateway between a SIP network and
   an XMPP network (a separate series of documents has been produced
   that defines such mappings).  More generally, this document does not
   describe service policies for inter-domain communication (often
   called "federation") between service providers (e.g., how a service
   provider that offers a combined SIP-XMPP service might communicate
   with a SIP-only or XMPP-only service), nor does it describe the
   reasons why a service provider might choose SIP or XMPP for various

   This document concentrates on use cases where the SIP services and
   XMPP services are controlled by one and the same provider, since that
   assumption greatly simplifies both client implementation and server-
   side deployment (e.g., a single service provider can enforce common
   or coordinated policies across both the SIP and XMPP aspects of a
   CUSAX service, which is not possible if a SIP service is offered by
   one provider and an XMPP service is offered by another).  Since this
   document is of an informational nature, it is not unreasonable for
   clients to apply some of the guidelines here even in cases where
   there is no established relationship between the SIP and the XMPP
   services (for example, it is reasonable for a client to provide a way
   for its users to easily start a call to a phone number recorded in a
   vCard).  However, the exact set of rules to follow in such cases is
   left to application developers.

   Finally, this document makes a further simplifying assumption by
   discussing only the use of a single client, not use of and
   coordination among multiple endpoints controlled by the same user
   (e.g., user agents running simultaneously on a laptop computer,
   tablet, and mobile phone).

2.  Client Bootstrap

   One of the main problems of using two distinct protocols when
   providing one service is the impact on usability.  Email services,
   for example, have long been affected by the mixed use of SMTP for
   outgoing mail and POP3 or IMAP for incoming mail.  Although standard
   service discovery methods (such as the proper DNS records) make it
   possible for a user agent to locate the right host(s) at which to
   connect, they do not provide the kind of detailed information that is

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   needed to actually configure the user agent for use with the service.
   As a result, it is rather complicated for inexperienced users to
   configure a mail client and start using it with a new service, and
   Internet service providers often need to provide configuration
   instructions for various mail clients.  Client developers and
   communication device manufacturers on the other hand often ship with
   a number of wizards that enable users to easily set up a new account
   for a number of popular email services.  While this may improve the
   situation to some extent, the user experience is still clearly sub-

   While it should be possible for CUSAX users to manually configure
   their separate SIP and XMPP accounts, dual-stack SIP/XMPP clients
   ought to provide means of online provisioning, typically by means of
   a web-based service at an HTTP URI.  While the specifics of such
   mechanisms are outside the scope of this specification, they should
   make it possible for a service provider to remotely configure the
   clients based on minimal user input (e.g., only a user ID and

   Because many of the features that a CUSAX client would privilege in
   one protocol would also be available in the other, clients should
   make it possible for such features to be disabled for a specific
   account.  In particular, it is suggested that clients allow for audio
   and video calling features to be disabled for XMPP accounts, and that
   instant messaging and presence features should also be made optional
   for SIP accounts.

   The main advantage of this approach is that clients would be able to
   continue to function properly and use the complete feature set of
   standalone SIP and XMPP accounts.

   Once clients have been provisioned, they need to independently log
   into the SIP and XMPP accounts that make up the CUSAX "service" and
   then maintain both these connections as displayed in Figure 2.

   | Provisioning |-----------+
   |    Server    |           |
   +--------------+           v
      |              +----------------+
      |              | vCard Storage/ |
      |              | User Directory |
      |              +----------------+
      |                /            \
      |      +------------+      +-------------+
      |      | SIP Server |      | XMPP Server |
      |      +------------+      +-------------+

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      |               \             /
      |      media     \           /  instant messaging,
      |      signaling  \         /   presence, etc.
      |                  \       /
      |               +--------------+
      +---------------| CUSAX Client |

                       Figure 2: Example Deployment

   In order to improve the user experience, when reporting connection
   status clients may also wish to present the XMPP connection as an
   "instant messaging" or a "chat" account.  Similarly they could also
   depict the SIP connection as a "Voice and Video" or a "Telephony"
   connection.  The exact naming is of course entirely up to
   implementers.  The point is that, in cases where SIP and XMPP are
   components of a service offered by a single provider, such
   presentation could help users better understand why they are being
   shown two different connections for what they perceive as a single
   service.  It could alleviate especially situations where one of these
   connections is disrupted while the other one is still active.
   Naturally, the developers of a CUSAX client or the providers of a
   CUSAX service might decide not to accept such situations and force a
   client to completely disconnect unless both aspects are successfully

   Clients may also choose to delay their XMPP connection until they
   have been successfully registered on SIP.  This would help avoid the
   situation where a user appears online to its contacts but calling it
   would fail because their clients is still connecting to the SIP
   aspect of their CUSAX service.

3.  Operation

   Once a CUSAX client has been provisioned and authorized to connect to
   the corresponding SIP and XMPP services it would proceed by
   retrieving its XMPP roster.

   The client should use XMPP for all forms of communication with the
   contacts from this roster, which will occur naturally because they
   were retrieved through XMPP.  Audio/video features however, are
   disabled in the XMPP stack, so any form of communication based on
   these features (e.g.  direct calls, conferences, desktop streaming,
   etc.)  will happen over SIP.  The rest of this section describes
   deployment, discovery, usability and linking semantics that allow
   CUSAX clients to fall back and seamlessly use SIP for these features.

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3.1.  Server-Side Setup

   In order for CUSAX to function properly, XMPP service administrators
   should make sure that at least one of the vCard [RFC6350] "tel"
   fields for each contact is properly populated with a SIP URI or a
   phone number when an XMPP protocol for vCard storage is used (e.g.,
   [XEP-0054] or [XEP-0292]).  There are no limitations as to the form
   of that number.  For example while it is desirable to maintain a
   certain consistency between SIP AORs and XMPP JIDs, that is by no
   means required.  It is quite important however that the phone number
   or SIP AOR stored in the vCard be reachable through the SIP aspect of
   this CUSAX service.

   Administrators may also choose to include the "video" tel type
   defined in [RFC6350] for accounts that would be capable of handling
   video communication.

   To ensure that the foregoing approach is always respected, service
   providers might consider (1) preventing clients (and hence users)
   from modifying the vCard "tel" fields or (2) applying some form of
   validation before storing changes.  Of course such validation would
   be feasible mostly in cases where a single provider controls both the
   XMPP and the SIP service since such providers would "know" (e.g.,
   based on use of a common user database for both services) what SIP
   AOR corresponds to a given XMPP user (as indicated in Figure 2).

3.2.  Client-Side Discovery and Usability

   When rendering the roster for a particular XMPP account CUSAX clients
   should make sure that users are presented with a "Call" option for
   each roster entry that has a properly set "tel" field.  This is the
   case even if calling features have been disabled for that particular
   XMPP account, as advised by this document.  The usefulness of such a
   feature is not limited to CUSAX.  After all, numbers are entered in
   vCards in order to be dialed and called.  Hence, as long as an XMPP
   client has any means of conducting a call it may wish to make it
   possible for the user to easily dial any numbers that it learned
   through whatever means.

   Clients that have separate triggers (buttons) for audio and video
   calls may choose to use the presence or absence of the "video" tel
   type defined in [RFC6350] and enable or disable the possibility for
   starting video calls accordingly.

   In addition to discovering phone numbers from vCards, clients may
   also check for alternative communication methods as advertised in
   XMPP presence broadcasts and Personal Eventing Protocol nodes as
   described in XEP-0152: Reachability Addresses [XEP-0152].  However,

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   these indications are merely hints, and a receiving client ought not
   associate a SIP address and an XMPP address unless it has some way to
   verify the association (e.g., the vCard of the XMPP account lists the
   SIP address and the vCard of the SIP account lists the XMPP address,
   or the association is made explicit in a record provided by a trusted
   directory).  Alternatively or in cases where vCard or directory data
   is not available, a CUSAX client could take the user's own address
   book as the canonical source for contact addresses.

3.3.  Indicating a Relation Between SIP and XMPP Accounts

   In order to improve usability, in cases where clients are provisioned
   with only a single telephony-capable account they ought to initiate
   calls immediately upon user request without asking users to indicate
   an account that the call should go through.  This way CUSAX users
   (whose only account with calling capabilities is usually the SIP part
   of their service) would have a better experience, since from the
   user's perspective calls "just work at the click of a button".

   In some cases however, clients will be configured with more than the
   two XMPP and SIP accounts provisioned by the CUSAX provider.  Users
   are likely to add additional stand-alone XMPP or SIP accounts (or
   accounts for other communications protocols), any of which might have
   both telephony and instant messaging capabilities.  Such situations
   can introduce additional ambiguity since all of the telephony-capable
   accounts could be used for calling the numbers the client has learned
   from the vCards.

   To avoid such confusion, client implementers and CUSAX service
   providers may choose to indicate the existence of a special
   relationship between the SIP and XMPP accounts of a CUSAX service.
   For example, let's say that Alice's service provider has opened both
   an XMPP account and a SIP account for her.  During or after
   provisioning, her client could indicate that
   has a CUSAX relation to (i.e., that they are
   two aspects of the same service).  This way whenever Alice triggers a
   call to a contact in her XMPP roster, the client would preferentially
   initiate this call through her SIP account even if other
   possibilities exist (such as the XMPP account where the vCard was
   obtained or a SIP account with another provider).

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   If, on the other hand, no relationship has been configured or
   discovered between a SIP account and an XMPP account, and the client
   is aware of multiple telephony-capable accounts, it ought to present
   the user with the choice of reaching the contact through any of those
   accounts.  This includes the source XMPP account where the vCard was
   obtained (in case its telephony capabilities are not disabled through
   configuration or provisioning), in order to guarantee proper
   operation for XMPP accounts that are not part of a CUSAX deployment.

3.4.  Matching Incoming SIP Calls to XMPP JIDs

   When receiving SIP calls, clients may wish to determine the identity
   of the caller and a corresponding XMPP roster entry so that users
   could revert to chatting or other forms of communication that require
   XMPP.  To do so clients could search their roster for an entry whose
   vCard has a "tel" field matching the originator of the call.

   In addition, in order to avoid the effort of iterating over an entire
   roster and retrieving all vCards, CUSAX clients may use a SIP Call-
   Info header whose 'purpose' header field parameter has a value of
   "impp" as described in [I-D.saintandre-impp-call-info].  An example

   Call-Info: <> ;purpose=impp

   Note that the information from the Call-Info header should only be
   used as a cue: the actual AOR-to-JID binding would still need to be
   confirmed by a vCard entry or through some other trusted means (such
   as an enterprise directory).  If this confirmation succeeds the
   client would not need to search the entire roster and retrieve all
   vCards.  Not performing the check might enable any caller (including
   malicious ones) to employ someone else's identity and perform various
   scams or Man-in-the-Middle attacks.

4.  Multi-Party Interactions

   CUSAX clients that support the SIP conferencing framework [RFC4353]
   can detect when a call they are participating in is actually a
   conference and can then subscribe for conference state updates as per
   [RFC4575].  A regular SIP user agent would also use the same
   conference URI for text communication with the Message Session Relay
   Protocol (MSRP).  However, given that SIP's instant messaging
   capabilities would normally be disabled (or simply not supported) in
   CUSAX deployments, an XMPP Multi-User Chat (MUC) [XEP-0045]
   associated with the conference can be announced/discovered through
   <service-uris> bearing the "grouptextchat" purpose
   [I-D.ivov-grouptextchat-purpose].  Similarly, an XMPP MUC can

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   advertise the SIP URI of an associated service for audio/video
   interactions using the 'audio-video-uri' field of the "muc#roominfo"
   data form [XEP-0004] to include extended information [XEP-0128] about
   the MUC room within XMPP service discovery [XEP-0030]; see [XEP-0045]
   for an example.

   Once a CUSAX client joins the MUC associated with a particular call
   it should not rely on any synchronization between the two.  Both the
   SIP conference and the XMPP MUC would function independently, each
   issuing and delivering its own state updates.  It is hence possible
   that that certain peers would temporarily or permanently be reachable
   in only one of the two conferences.  This would typically be the case
   with single-stack clients that have only joined the SIP call or the
   XMPP MUC.  It is therefore important for CUSAX clients to provide a
   clear indication to users as to the level of participation of the
   various participants.  In other words, a user needs to be able to
   easily understand whether a certain participant can receive text
   messages, audio/video, or both.

   Of course, tighter integration between the XMPP MUC and the SIP
   conference is also possible.  Permissions, roles, kicks and bans that
   are granted and performed in the MUC can easily be imitated by the
   conference focus/mixer into the SIP call.  If for example, a certain
   MUC member is muted, the conference mixer can choose to also apply
   the mute on the media stream corresponding to that participant.  The
   details and exact level of such integration is of course entirely up
   to implementers and service providers.

   The approach above describes one relatively lightweight possibility
   of combining SIP and XMPP multi-party interaction semantics without
   requiring tight integration between the two.  As with the rest of
   this specification, this approach is by no means normative.
   Implementation and future specifications may define other methods or
   provide other suggestions for improving the Unified Communications
   user experience in cases of multi-user chats in conference calling.

5.  Federation

   In theory there are no technical reasons why federation would require
   special behaviour from CUSAX clients.  However, it is worth noting
   that differences in administration policies may sometimes lead to
   potentially confusing user experiences.

   For example, let's say observes the CUSAX
   policies described in this specification.  All XMPP users at are hence configured to have vCards that match
   their SIP identities.  Alice is therefore used to making free, high-
   quality SIP calls to all the people in her roster.  Alice can also

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   make calls to the PSTN by simply dialing numbers.  She may even be
   used to these calls being billed to her online account so she would
   careful about how long they last.  This is not a problem for her
   since she can easily distinguish between a free SIP call (one that
   she made by calling one her roster entries) from a paid PSTN call
   that she dialed as a number.

   Then Alice adds  The Biloxi domain only
   has an XMPP service.  There is no SIP server and Bob uses a regular,
   XMPP-only client.  Bob has however added his mobile number to his
   vCard in order to make it easily accessible to his contacts.  Alice's
   client would pick up this number and make it possible for Alice to
   start a call to Bob's mobile phone number.

   This could be a problem because, other than the fact that Bob's
   address is from a different domain, Alice would have no obvious and
   straightforward cues telling her that this is in fact a call to the
   PSTN.  In addition to the potentially lower audio quality, Alice may
   also end up incurring unexpected charges for such calls.

   In order to avoid such issues, providers maintaining a CUSAX service
   for the users in their domain may choose to provide additional cues
   (e.g., a user interface warning or an an audio tone or message)
   indicating that a call would incur charges.

   A slightly less disturbing scenario, where a SIP service might only
   allow communication with intra-domain numbers, would simply prevent
   Alice from establishing a call with Bob's mobile.  Providers should
   hence make sure that calls to extra-domain numbers are flagged with
   an appropriate audio or textual warning.

6.  Security Considerations

   Use of the same user agent with two different accounts providing
   complementary features introduces the possibility of mismatches
   between the security profiles of those accounts or features.  For
   example, the SIP aspect and XMPP aspect of the CUSAX service might
   offer different authentication options (e.g., digest authentication
   for SIP as specified in [RFC3261] and SCRAM authentication [RFC5802]
   for XMPP as specified in [RFC6120]).  Similarly, a CUSAX client might
   successfully negotiate Transport Layer Security (TLS) [RFC5246] when
   connecting to the XMPP aspect of the service but not when connecting
   to the SIP aspect.  Such mismatches could introduce the possibility
   of downgrade attacks.  User agent developers and service providers
   ought to ensure that such mismatches are avoided as much as possible.

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   Refer to the specifications for the relevant SIP and XMPP features
   for detailed security considerations applying to each "stack" in a
   CUSAX client.

7.  IANA Considerations

   This document has no actions for the IANA.

8.  Informative References

              Ivov, E., "A Group Text Chat Purpose for Conference and
              Service URIs in the Session Initiation Protocol (SIP)
              Event Package for Conference State", draft-ivov-
              grouptextchat-purpose-00 (work in progress), April 2013.

              Saint-Andre, P., "Instant Messaging and Presence Purpose
              for the Call-Info Header in the Session Initiation
              Protocol (SIP)", draft-saintandre-impp-call-info-02 (work
              in progress), April 2013.

   [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.

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353, February

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
              Initiation Protocol (SIP) Event Package for Conference
              State", RFC 4575, August 2006.

   [RFC4825]  Rosenberg, J., "The Extensible Markup Language (XML)
              Configuration Access Protocol (XCAP)", RFC 4825, May 2007.

   [RFC4975]  Campbell, B., Mahy, R., and C. Jennings, "The Message
              Session Relay Protocol (MSRP)", RFC 4975, September 2007.

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

   [RFC5802]  Newman, C., Menon-Sen, A., Melnikov, A., and N. Williams,
              "Salted Challenge Response Authentication Mechanism
              (SCRAM) SASL and GSS-API Mechanisms", RFC 5802, July 2010.

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

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

   [RFC6350]  Perreault, S., "vCard Format Specification", RFC 6350,
              August 2011.

              Eatmon, R., Hildebrand, J., Miller, J., Muldowney, T., and
              P. Saint-Andre, "Data Forms", XSF XEP 0004, August 2007.

              Hildebrand, J., Millard, P., Eatmon, R., and P. Saint-
              Andre, "Service Discovery", XSF XEP 0030, June 2008.

              Saint-Andre, P., "Multi-User Chat", XSF XEP 0045, February

              Saint-Andre, P., "vcard-temp", XSF XEP 0054, July 2008.

              Saint-Andre, P., "Service Discovery Extensions", XSF XEP
              0128, October 2004.

              Hildebrand, J. and P. Saint-Andre, "XEP-0152: Reachability
              Addresses", XEP XEP-0152, February 2013.

              Saint-Andre, P. and S. Mizzi, "vCard4 Over XMPP", XSF XEP
              0292, October 2011.

Appendix A.  Acknowledgements

   This draft is inspired by the "SIXPAC" work of Markus Isomaki and
   Simo Veikkolainen.  Markus also provided various suggestions for
   improving the document.

   The authors would also like to thank the following persons for their
   reviews and suggestions: Sebastien Couture, Richard Brady, Olivier
   Crete, Aaron Evans, Kevin Gallagher, Adrian Georgescu, Saul Ibarra
   Corretge, David Laban, Murray Mar, Daniel Pocock, Travis Reitter, and
   Gonzalo Salgueiro.

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Authors' Addresses

   Emil Ivov
   Strasbourg  67000

   Phone: +33-672-811-555

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

   Phone: +1-303-308-3282

   Enrico Marocco
   Telecom Italia
   Via G. Reiss Romoli, 274
   Turin  10148


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