<pre>Internet Engineering Task Force (IETF)                           E. Ivov
Request for Comments: 7081                                         Jitsi
Category: Informational                                   P. Saint-Andre
ISSN: 2070-1721                                      Cisco Systems, Inc.
                                                              E. Marocco
                                                          Telecom Italia
                                                           November 2013


      <span class="h1">CUSAX: Combined Use of the Session Initiation Protocol (SIP)</span>
       <span class="h1">and the Extensible Messaging and Presence Protocol (XMPP)</span>

Abstract

   This document suggests some strategies for the combined use of the
   Session Initiation Protocol (SIP) and the Extensible Messaging and
   Presence Protocol (XMPP) both in user-oriented clients and in
   deployed servers.  Such strategies, which mainly consist of
   configuration changes and minimal software modifications to existing
   clients and servers, aim to provide a single, full-featured, real-
   time communication service by using complementary subsets of features
   from SIP and from XMPP.  Typically, such subsets consist of telephony
   capabilities from SIP and instant messaging and presence capabilities
   from XMPP.  This document does not define any new protocols or syntax
   for either SIP or XMPP and, by intent, does not attempt to
   standardize "best current practices".  Instead, it merely aims to
   provide practical guidance to those who are interested in the
   combined use of SIP and XMPP for real-time communication.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see <a href="./rfc5741#section-2">Section&nbsp;2 of RFC 5741</a>.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   <a href="http://www.rfc-editor.org/info/rfc7081">http://www.rfc-editor.org/info/rfc7081</a>.







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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 <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (<a href="http://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>) 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

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-2">2</a>
   <a href="#section-2">2</a>. Client Bootstrap ................................................<a href="#page-5">5</a>
   <a href="#section-3">3</a>. Operation .......................................................<a href="#page-6">6</a>
      <a href="#section-3.1">3.1</a>. Server-Side Setup ..........................................<a href="#page-7">7</a>
      <a href="#section-3.2">3.2</a>. Service Management .........................................<a href="#page-7">7</a>
      <a href="#section-3.3">3.3</a>. Client-Side Discovery and Usability ........................<a href="#page-8">8</a>
      <a href="#section-3.4">3.4</a>. Indicating a Relationship between SIP and XMPP Accounts ....<a href="#page-9">9</a>
      <a href="#section-3.5">3.5</a>. Matching Incoming SIP Calls to XMPP JIDs ..................<a href="#page-10">10</a>
   <a href="#section-4">4</a>. Multi-Party Interactions .......................................<a href="#page-11">11</a>
   <a href="#section-5">5</a>. Federation .....................................................<a href="#page-12">12</a>
   <a href="#section-6">6</a>. Summary of Suggested Strategies ................................<a href="#page-13">13</a>
   <a href="#section-7">7</a>. Security Considerations ........................................<a href="#page-14">14</a>
   <a href="#section-8">8</a>. References .....................................................<a href="#page-15">15</a>
      <a href="#section-8.1">8.1</a>. Normative References ......................................<a href="#page-15">15</a>
      <a href="#section-8.2">8.2</a>. Informative References ....................................<a href="#page-16">16</a>
   <a href="#appendix-A">Appendix A</a>. Acknowledgements ......................................<a href="#page-18">18</a>

<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   Historically, SIP [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] and XMPP [<a href="./rfc6120" title="&quot;Extensible Messaging and Presence Protocol (XMPP): Core&quot;">RFC6120</a>] 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".  On the other hand, XMPP has, from its
   Jabber days, been mostly used for instant messaging, presence
   [<a href="./rfc6121" title="&quot;Extensible Messaging and Presence Protocol (XMPP): Instant Messaging and Presence&quot;">RFC6121</a>], and related services such as groupchat rooms [<a href="#ref-XEP-0045" title="&quot;Multi-User Chat&quot;">XEP-0045</a>].









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   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  In the context of the SIP for Instant Messaging and Presence
      Leveraging Extensions (SIMPLE) working group, the IETF has defined
      a number of protocols and protocol extensions that not only allow
      for SIP to be used for regular instant messaging and presence but
      that also provide mechanisms for related features such as
      multi-party chat, server-stored contact lists, and file transfer
      [<a href="./rfc6914" title="&quot;SIMPLE Made Simple: An Overview of the IETF Specifications for Instant Messaging and Presence Using the Session Initiation Protocol (SIP)&quot;">RFC6914</a>].

   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 [<a href="#ref-XEP-0166" title="&quot;Jingle&quot;">XEP-0166</a>], and arguably their most
      popular use case is audio and video calling [<a href="#ref-XEP-0167" title="&quot;Jingle RTP Sessions&quot;">XEP-0167</a>].

   However, although SIP has been extended for messaging and presence
   and XMPP has been extended for voice and video, the reality is that
   SIP remains the protocol of choice for telephony-like services, and
   XMPP remains the protocol of choice for IM and presence services.  As
   a result, a number of adopters have found themselves needing features
   that are not offered by any single-protocol solution, but ones that
   separately exist in SIP and XMPP implementations.  The idea of
   seamlessly using both protocols together would hence often appeal to
   service providers and users.  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





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   This document suggests different configuration options and minimal
   modifications to existing software so that clients and servers can
   offer these hybrid services while providing an optimal user
   experience.  It covers server discovery, determining a SIP Address of
   Record (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 we also suggest certain server-side
   configurations and operational strategies.  The document also
   discusses significant security considerations that can arise when
   offering a dual-protocol solution and provides advice for avoiding
   security mismatches that would result in degraded communications
   security for end users.

   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 CUSAX 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 features.

   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 provider).
   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
   or SIP URI found in a vCard or obtained from a user directory).
   However, the strategies to pursue in such cases are left to
   application developers.

   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).  Although user agents running on separate endpoints might
   themselves be CUSAX clients or might engage in different aspects of
   an interaction (e.g., a user might employ her mobile phone for audio



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   and her tablet for video and text chat), such usage complicates the
   guidelines for developers of user agents and therefore is left as a
   matter of implementation for now.

   It is important to note that this document does not attempt to
   standardize "best current practices" in the sense defined in the
   Internet Standards Process [<a href="./rfc2026" title="&quot;The Internet Standards Process -- Revision 3&quot;">RFC2026</a>].  Instead, it collects together
   informational documentation about some strategies that might prove
   helpful to those who implement and deploy combined SIP/XMPP software
   and services.  With sufficient use and appropriate modification to
   incorporate the lessons of experience, these strategies might someday
   form the basis for standardization of best current practices.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Client Bootstrap</span>

   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 Post Office Protocol version 3 (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) for connect purposes, they do not provide the kind
   of detailed information that is 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 as a result, 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
   so-called "wizard" interfaces that enable users to easily configure
   accounts with a number of popular email services.  Although this may
   improve the situation to some extent, the user experience is still
   clearly suboptimal.

   While it should be possible for CUSAX users to manually configure
   their separate SIP and XMPP accounts (often using "wizards"), service
   providers offering CUSAX services to users of dual-stack SIP/XMPP
   clients ought to provide methods for online provisioning, typically
   by means of a web-based service at an HTTPS URL (naturally, single-
   purpose SIP services or XMPP services could offer such methods as
   well, but they can be especially helpful where the two aspects of the
   CUSAX service need to have several configuration options in common).
   Although the specifics of such mechanisms are outside the scope of
   this document, 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 password).  As far as the authors are aware, no
   open protocol for endpoint configuration is yet available and




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   adopted; however, application developers are encouraged to explore
   the potential for future progress in this space (e.g., perhaps based
   on technologies such as WebFinger [<a href="./rfc7033" title="&quot;WebFinger&quot;">RFC7033</a>]).

   By default, when a CUSAX client is used in concert with SIP and XMPP
   accounts that have a CUSAX relationship (see <a href="#section-3.4">Section 3.4</a>), the client
   should disable audio and video calling over XMPP and disable instant
   messaging and presence over SIP.  (It is a matter of implementation
   whether a CUSAX client allows a user to override these defaults in
   various ways, e.g., by domain, by individual contact, or by device.)
   The main advantage of this approach is that a client would employ the
   most relevant features from both SIP and XMPP when used in the
   context of a CUSAX service.  Note that this default configuration
   does not apply to stand-alone SIP accounts or XMPP accounts, for
   which other settings are likely to be more appropriate (see
   <a href="#section-3.4">Section 3.4</a> for details).

   Once a client has been provisioned, it needs to independently log
   into the SIP account and XMPP account that make up the CUSAX
   "service" and then maintain both connections.

   In order to improve the user experience, when reporting connection
   status, a CUSAX client may wish to present the XMPP connection as an
   "instant messaging" or a "chat" account and 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 (especially when one of the connections is
   disrupted while the other one is still active).  Alternatively, the
   developers of a CUSAX client or the providers of a CUSAX service
   might decide to force a client to completely disconnect unless both
   aspects are successfully connected.

   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 her contacts but calling the
   user's client would fail because the user's client is still
   connecting to the SIP aspect of the CUSAX service.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Operation</span>

   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.





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

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  Server-Side Setup</span>

   In order for CUSAX to function properly, XMPP service administrators
   should make sure that at least one of the vCard [<a href="./rfc6350" title="&quot;vCard Format Specification&quot;">RFC6350</a>] "tel"
   fields for each contact is properly populated with a SIP URI for the
   user's address at the SIP audio/video service provided by the CUSAX
   server.  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.
   (The same considerations apply even if the directory storage format
   is not vCard storage over XMPP as described by [<a href="#ref-XEP-0054" title="&quot;vcard-temp&quot;">XEP-0054</a>] or
   [<a href="#ref-XEP-0292" title="&quot;vCard4 Over XMPP&quot;">XEP-0292</a>].)

   Administrators may also choose to include the "video" tel type
   defined in [<a href="./rfc6350" title="&quot;vCard Format Specification&quot;">RFC6350</a>] for accounts that would be capable of handling
   video communication.

   To ensure that the foregoing approach is always respected, service
   providers might consider validating the values of vCard "tel" fields
   before storing changes.  Of course, such validation would be feasible
   only 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.

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  Service Management</span>

   The task of operating and managing a stand-alone SIP service or XMPP
   service is not always easy.  Combining the two into a unified service
   introduces additional challenges, including:

   o  The necessity of opening additional ports on the client side if
      SIP functionality is added to an existing XMPP deployment, or vice
      versa.





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   o  The potential for important differences in security posture across
      SIP and XMPP (e.g., SIP servers and XMPP servers might support
      different Transport Layer Security (TLS) ciphersuites).

   o  The need for, ideally, a common authentication backend and other
      infrastructure that is shared across the SIP and XMPP aspects of
      the combined service.

   o  Coordinated monitoring and logging of the SIP and XMPP servers to
      enable the correlation of incidents and the pinpointing of
      problems.

   o  The difficulty of troubleshooting client-side issues, e.g., if the
      client loses connectivity for XMPP but maintains its SIP
      connection.

   Although separation of functionality (SIP for media and XMPP for IM
   and presence) can help to ease the operational burden to some extent,
   service providers are urged to address the foregoing challenges and
   similar issues when preparing to launch a CUSAX service.

   Beyond the issues listed above, service providers might want to be
   aware of more subtle operational issues that can arise.  For example,
   if a service provider uses different network operators for the SIP
   service and the XMPP service, end-to-end connectivity might be more
   reliable or consistent in one service than in the other service.
   Similar issues can arise when the media path and the signaling path
   go over different networks, even in stand-alone SIP or XMPP services.
   Providers of CUSAX services are advised to consider the potential for
   such topologies to cause operational challenges.

<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  Client-Side Discovery and Usability</span>

   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 or stored in directories 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 (e.g., buttons) for audio calls
   and video calls may choose to use the presence or absence of the
   "video" tel type defined in [<a href="./rfc6350" title="&quot;vCard Format Specification&quot;">RFC6350</a>] as the basis for choosing




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   whether to enable or disable the possibility for starting video calls
   (i.e., if there is no "video" tel type for a particular contact, the
   client could disable the "video call" button for that contact).

   In addition to discovering phone numbers from vCards or user
   directories, 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" [<a href="#ref-XEP-0152" title="&quot;XEP-0152: Reachability Addresses&quot;">XEP-0152</a>].  However, 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 relationship (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 relationship 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.

<span class="h3"><a class="selflink" id="section-3.4" href="#section-3.4">3.4</a>.  Indicating a Relationship between SIP and XMPP Accounts</span>

   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 vCards or directories.

   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 alice@xmpp.example.com
   has a CUSAX relationship to alice@sip.example.com (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 example.com SIP account
   even if other possibilities exist (such as the XMPP account where the



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   vCard was obtained or a SIP account with another provider).
   Similarly, the client would preferentially initiate textual chat
   sessions using her XMPP account.

   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 option of using XMPP Jingle as one method for
   engaging in audio and video interactions with a contact who has an
   XMPP address.  This can help to ensure that a CUSAX user can complete
   audio and video calls with XMPP users who are not part of a CUSAX
   deployment.

<span class="h3"><a class="selflink" id="section-3.5" href="#section-3.5">3.5</a>.  Matching Incoming SIP Calls to XMPP JIDs</span>

   When receiving a SIP call, a CUSAX client may wish to determine the
   identity of the caller and a corresponding XMPP roster entry so that
   the receiving user could revert to chatting or other forms of
   communication that require XMPP.  To do so, a CUSAX client could
   search the user's 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 the entire roster of the user and
   retrieving vCards for all of the user's contacts, the receiving
   client may guess at the identity of the caller based a SIP Call-Info
   header whose 'purpose' header field parameter has a value of "impp"
   as described in [<a href="./rfc6993" title="&quot;Instant Messaging and Presence Purpose for the Call-Info Header Field in the Session Initiation Protocol (SIP)&quot;">RFC6993</a>].  To enable this usage, a sending client
   would need to include such a Call-Info header in the SIP messages
   that it sends when initiating a call.  An example follows.

   Call-Info: &lt;xmpp:alice@xmpp.example.com&gt; ;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 the vCard of a contact in the receiving user's roster 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.

   However, although an AOR-to-JID binding can be a helpful hint to the
   user, nothing in the foregoing paragraph ought to be construed as
   necessarily discouraging users, clients, or service providers from
   accepting calls originated by entities that are not established
   contacts of the user (e.g., as reflected in the user's roster); that
   is a policy matter for the user, client, or service provider.




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   It is also worth noting that callers preferring to remain anonymous
   as per [<a href="./rfc3325" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">RFC3325</a>] would not provide Call-Info information.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Multi-Party Interactions</span>

   CUSAX clients that support the SIP conferencing framework [<a href="./rfc4353" title="&quot;A Framework for Conferencing with the Session Initiation Protocol (SIP)&quot;">RFC4353</a>]
   can detect when a call they are participating in is actually a
   conference and can then subscribe to conference state updates as per
   [<a href="./rfc4575" title="&quot;A Session Initiation Protocol (SIP) Event Package for Conference State&quot;">RFC4575</a>].  A regular SIP user agent might 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) room [<a href="#ref-XEP-0045" title="&quot;Multi-User Chat&quot;">XEP-0045</a>]
   associated with the conference can be announced/discovered through
   &lt;service-uris&gt; bearing the "grouptextchat" purpose [<a href="#ref-GROUPTEXTCHAT" title="&quot;A Group Text Chat Purpose for Conference and Service URIs in the Session Initiation Protocol (SIP) Event Package for Conference State&quot;">GROUPTEXTCHAT</a>].
   Similarly, an XMPP MUC room can advertise the SIP URI of an
   associated service for audio/video interactions using the
   'audio-video-uri' field of the "muc#roominfo" data form [<a href="#ref-XEP-0004" title="&quot;Data Forms&quot;">XEP-0004</a>] to
   include extended information [<a href="#ref-XEP-0128" title="&quot;Service Discovery Extensions&quot;">XEP-0128</a>] about the MUC room within
   XMPP service discovery [<a href="#ref-XEP-0030" title="&quot;Service Discovery&quot;">XEP-0030</a>]; see [<a href="#ref-XEP-0045" title="&quot;Multi-User Chat&quot;">XEP-0045</a>] for an example.
   These methods would enable a CUSAX-aware SIP conference server to
   advertise the existence of an associated XMPP chat room and for a
   CUSAX-aware XMPP chat room to advertise the existence of an
   associated SIP conference server.

   If a CUSAX client joins the MUC room associated with a particular
   call, it should not rely on any synchronization between the two.
   Both the SIP conference and the XMPP MUC room would function
   independently, each issuing and delivering its own state updates.
   Hence, it is possible 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 room.  It is therefore important
   for CUSAX clients to provide a clear indication to users as to the
   level of involvement of the various participants: i.e., a user needs
   to be able to easily understand whether a certain participant can
   receive text messages, audio/video, or both.

   At the level of the CUSAX service, it is also possible to enforce
   tighter integration between the XMPP MUC room and the SIP conference.
   Permissions, roles, kicks, and bans that are granted and performed in
   the MUC room 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.  However, the details and
   exact level of such integration are entirely up to implementers and
   service providers.




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   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 document, this approach is by no means normative.
   Implementations and future documents may define other methods or
   provide other suggestions for improving the unified communications
   user experience in cases of multi-user chats and conference calling.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Federation</span>

   In theory, there are no technical reasons why federation (i.e.,
   inter-domain communication) would require special behavior 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 atlanta.example.com observes the CUSAX
   policies described in this document.  All XMPP users at
   atlanta.example.com 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
   make calls to the Public Switched Telephone Network (PSTN) by simply
   dialing numbers.  She may even be used to these calls being billed to
   her online account, so she would be 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 of her
   roster entries) from a paid PSTN call that she dialed as a number.

   Then, Alice adds xmpp:bob@biloxi.example.com.  The Biloxi domain only
   has an XMPP service.  There is no SIP server and Bob uses an
   XMPP-only client.  However, Bob has 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 service-generated signal that triggers a user-interface
   warning in a CUSAX client, an auditory tone, or a spoken message)
   indicating that a call would incur unexpected charges.





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   Another scenario arises when a SIP service allows communication only
   with intra-domain numbers; here, Alice might be prevented from
   establishing a call with Bob's mobile phone.  Providers should
   therefore make sure that calls to inter-domain numbers are flagged
   with an appropriate audio or textual warning.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Summary of Suggested Strategies</span>

   The following strategies are suggested for CUSAX user agents:

   1.   By default, prefer SIP for audio and video and XMPP for
        messaging and presence.

   2.   Use XMPP for all forms of communication with the contacts from
        the XMPP roster, with the exception of features that are based
        on establishing real-time sessions (e.g., audio/video calls) for
        which SIP should be used.

   3.   Provide online provisioning options for providers to remotely
        set up SIP and XMPP accounts so that users wouldn't need to go
        through a multi-step configuration process.

   4.   Provide online provisioning options for providers to completely
        disable features for an account associated with a given protocol
        (SIP or XMPP) if the features are preferred in another protocol
        (XMPP or SIP).

   5.   Present a "Call" option for each roster entry that has a
        properly set "tel" field in the vCard or equivalent.

   6.   If the client is provisioned with only a single telephony-
        capable account, initiate calls immediately upon user request
        without asking users to indicate an account that the call should
        go through.

   7.   If 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, present the user with the
        choice of reaching the contact through any of those accounts.

   8.   If known, indicate the existence of a special relationship
        between the SIP and XMPP accounts of a CUSAX service.

   9.   Optionally, present the XMPP connection as an "instant
        messaging" or a "chat" account and the SIP connection as a
        "Voice and Video" or a "Telephony" account.





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   10.  Optionally, determine the identity of the audio/video caller and
        a corresponding XMPP roster entry so that the user could use
        textual chatting or other forms of communication that require
        XMPP.

   11.  Optionally, delay the XMPP connection until after a SIP
        connection has been successfully registered.

   12.  Optionally, check for alternative communication methods (SIP
        addresses advertised over XMPP and XMPP addresses advertised
        over SIP).

   The following strategies are suggested for CUSAX services:

   1.  Use online provisioning and configuration of accounts so that
       users won't need to set up two separate accounts for the CUSAX
       service.

   2.  Use online provisioning so that calling features are disabled for
       all XMPP accounts.

   3.  Ensure that at least one of the vCard "tel" fields for each XMPP
       user is properly populated with a SIP URI that is reachable
       through the SIP service.

   4.  Optionally, include the "video" tel type for accounts that are
       capable of handling video communication.

   5.  Optionally, provision clients with information indicating that
       specific SIP and XMPP accounts are related in a CUSAX service.

   6.  Optionally, attach a "Call-Info" header with an "impp" purpose to
       all SIP INVITE messages, so that clients can more rapidly
       associate a caller with a roster entry and display a "Caller ID".

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Security Considerations</span>

   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.  Two
   security mismatches of particular concern are:

   o  The SIP aspect and XMPP aspect of a CUSAX service might offer
      different authentication options (e.g., digest authentication for
      SIP as specified in [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] and Salted Challenge Response
      Authentication Mechanism (SCRAM) authentication [<a href="./rfc5802" title="&quot;Salted Challenge Response Authentication Mechanism (SCRAM) SASL and GSS-API Mechanisms&quot;">RFC5802</a>] for XMPP
      as specified in [<a href="./rfc6120" title="&quot;Extensible Messaging and Presence Protocol (XMPP): Core&quot;">RFC6120</a>]).  Because SIP uses a password-based
      method (digest) and XMPP uses a pluggable framework for



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      authentication via the Simple Authentication and Security Layer
      (SASL) technology [<a href="./rfc4422" title="&quot;Simple Authentication and Security Layer (SASL)&quot;">RFC4422</a>], it is also possible that the XMPP
      connection could be authenticated using a password-free method
      such as client certificates with SASL EXTERNAL, even though a
      username and password is used for the SIP connection.

   o  The Transport Layer Security (TLS) [<a href="./rfc5246" title="&quot;The Transport Layer Security (TLS) Protocol Version 1.2&quot;">RFC5246</a>] ciphersuites offered
      or negotiated on the XMPP side might be different from those on
      the SIP side because of implementation or configuration
      differences between the SIP server and the XMPP server.  Even more
      seriously, a CUSAX client might successfully negotiate TLS when
      connecting to the XMPP aspect of the service but not when
      connecting to the SIP aspect, or vice versa.  In this situation,
      an end user might think that the combined CUSAX session with the
      service is protected by TLS, even though only one aspect is
      protected.

   Security mismatches such as these (as well as others related to end-
   to-end encryption of messages or media) introduce the possibility of
   downgrade attacks, eavesdropping, information leakage, and other
   security vulnerabilities.  User agent developers and service
   providers must ensure that such mismatches are avoided as much as
   possible (e.g., by enforcing common and strong security
   configurations and policies across protocols).  Specifically, if both
   protocols are not safeguarded by similar levels of cryptographic
   protection, the user must be informed of that fact and given the
   opportunity to bring both up to the same level.

   <a href="#section-5">Section 5</a> discusses potential issues that may arise due to a mismatch
   between client capabilities, such as calls being initiated with costs
   that are not expected by the end user.  Such issues could be
   triggered maliciously, as well as by accident.  Implementers
   therefore need to provide necessary cues to raise user awareness as
   suggested in <a href="#section-5">Section 5</a>.

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

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  References</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  Normative References</span>

   [<a id="ref-RFC3261">RFC3261</a>]        Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                    Johnston, A., Peterson, J., Sparks, R., Handley, M.,
                    and E. Schooler, "SIP: Session Initiation Protocol",
                    <a href="./rfc3261">RFC 3261</a>, June 2002.




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   [<a id="ref-RFC6120">RFC6120</a>]        Saint-Andre, P., "Extensible Messaging and Presence
                    Protocol (XMPP): Core", <a href="./rfc6120">RFC 6120</a>, March 2011.

   [<a id="ref-RFC6121">RFC6121</a>]        Saint-Andre, P., "Extensible Messaging and Presence
                    Protocol (XMPP): Instant Messaging and Presence",
                    <a href="./rfc6121">RFC 6121</a>, March 2011.

<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  Informative References</span>

   [<a id="ref-GROUPTEXTCHAT">GROUPTEXTCHAT</a>]  Ivov, E., "A Group Text Chat Purpose for Conference
                    and Service URIs in the Session Initiation Protocol
                    (SIP) Event Package for Conference State", Work
                    in Progress, June 2013.

   [<a id="ref-RFC2026">RFC2026</a>]        Bradner, S., "The Internet Standards Process --
                    Revision 3", <a href="https://www.rfc-editor.org/bcp/bcp9">BCP 9</a>, <a href="./rfc2026">RFC 2026</a>, October 1996.

   [<a id="ref-RFC3325">RFC3325</a>]        Jennings, C., Peterson, J., and M. Watson, "Private
                    Extensions to the Session Initiation Protocol (SIP)
                    for Asserted Identity within Trusted Networks",
                    <a href="./rfc3325">RFC 3325</a>, November 2002.

   [<a id="ref-RFC4353">RFC4353</a>]        Rosenberg, J., "A Framework for Conferencing with
                    the Session Initiation Protocol (SIP)", <a href="./rfc4353">RFC 4353</a>,
                    February 2006.

   [<a id="ref-RFC4422">RFC4422</a>]        Melnikov, A. and K. Zeilenga, "Simple Authentication
                    and Security Layer (SASL)", <a href="./rfc4422">RFC 4422</a>, June 2006.

   [<a id="ref-RFC4575">RFC4575</a>]        Rosenberg, J., Schulzrinne, H., and O. Levin, "A
                    Session Initiation Protocol (SIP) Event Package for
                    Conference State", <a href="./rfc4575">RFC 4575</a>, August 2006.

   [<a id="ref-RFC5246">RFC5246</a>]        Dierks, T. and E. Rescorla, "The Transport Layer
                    Security (TLS) Protocol Version 1.2", <a href="./rfc5246">RFC 5246</a>,
                    August 2008.

   [<a id="ref-RFC5802">RFC5802</a>]        Newman, C., Menon-Sen, A., Melnikov, A., and N.
                    Williams, "Salted Challenge Response Authentication
                    Mechanism (SCRAM) SASL and GSS-API Mechanisms",
                    <a href="./rfc5802">RFC 5802</a>, July 2010.

   [<a id="ref-RFC6350">RFC6350</a>]        Perreault, S., "vCard Format Specification",
                    <a href="./rfc6350">RFC 6350</a>, August 2011.







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   [<a id="ref-RFC6914">RFC6914</a>]        Rosenberg, J., "SIMPLE Made Simple: An Overview of
                    the IETF Specifications for Instant Messaging and
                    Presence Using the Session Initiation Protocol
                    (SIP)", <a href="./rfc6914">RFC 6914</a>, April 2013.

   [<a id="ref-RFC6993">RFC6993</a>]        Saint-Andre, P., "Instant Messaging and Presence
                    Purpose for the Call-Info Header Field in the
                    Session Initiation Protocol (SIP)", <a href="./rfc6993">RFC 6993</a>,
                    July 2013.

   [<a id="ref-RFC7033">RFC7033</a>]        Jones, P., Salgueiro, G., Jones, M., and J. Smarr,
                    "WebFinger", <a href="./rfc7033">RFC 7033</a>, September 2013.

   [<a id="ref-XEP-0004">XEP-0004</a>]       Eatmon, R., Hildebrand, J., Miller, J., Muldowney,
                    T., and P. Saint-Andre, "Data Forms", XSF XEP 0004,
                    August 2007.

   [<a id="ref-XEP-0030">XEP-0030</a>]       Hildebrand, J., Millard, P., Eatmon, R., and P.
                    Saint-Andre, "Service Discovery", XSF XEP 0030,
                    June 2008.

   [<a id="ref-XEP-0045">XEP-0045</a>]       Saint-Andre, P., "Multi-User Chat", XSF XEP 0045,
                    February 2012.

   [<a id="ref-XEP-0054">XEP-0054</a>]       Saint-Andre, P., "vcard-temp", XSF XEP 0054,
                    July 2008.

   [<a id="ref-XEP-0128">XEP-0128</a>]       Saint-Andre, P., "Service Discovery Extensions", XSF
                    XEP 0128, October 2004.

   [<a id="ref-XEP-0152">XEP-0152</a>]       Hildebrand, J. and P. Saint-Andre, "XEP-0152:
                    Reachability Addresses", XEP XEP-0152,
                    September 2013.

   [<a id="ref-XEP-0166">XEP-0166</a>]       Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R.,
                    Egan, S., and J. Hildebrand, "Jingle", XSF XEP 0166,
                    December 2009.

   [<a id="ref-XEP-0167">XEP-0167</a>]       Ludwig, S., Saint-Andre, P., Egan, S., McQueen, R.,
                    and D. Cionoiu, "Jingle RTP Sessions", XSF XEP 0167,
                    December 2009.

   [<a id="ref-XEP-0292">XEP-0292</a>]       Saint-Andre, P. and S. Mizzi, "vCard4 Over XMPP",
                    XSF XEP 0292, September 2013.







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<span class="h2"><a class="selflink" id="appendix-A" href="#appendix-A">Appendix A</a>.  Acknowledgements</span>

   This document 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 people for their
   reviews and suggestions: Sebastien Couture, Dan-Christian Bogos,
   Richard Brady, Olivier Crete, Aaron Evans, Kevin Gallagher, Adrian
   Georgescu, Saul Ibarra Corretge, David Laban, Gergely Lukacsy,
   Spencer MacDonald, Murray Mar, Daniel Pocock, Travis Reitter, and
   Gonzalo Salgueiro.

   Brian Carpenter, Ted Hardie, Paul Hoffman, and Benson Schliesser
   reviewed the document on behalf of the General Area Review Team, the
   Applications Area Directorate, the Security Directorate, and the
   Operations and Management Directorate, respectively.

   Benoit Claise, Barry Leiba, and Pete Resnick provided helpful and
   substantive feedback during IESG review.

   The document shepherd was Mary Barnes.  The sponsoring Area Director
   was Gonzalo Camarillo.




























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Authors' Addresses

   Emil Ivov
   Jitsi
   Strasbourg  67000
   France

   Phone: +33-177-624-330
   EMail: emcho@jitsi.org


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

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


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

   EMail: enrico.marocco@telecomitalia.it























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