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Network Working Group                                            E. Ivov
Internet-Draft                                                     Jitsi
Intended status: Informational                            P. Saint-Andre
Expires: April 02, 2014                              Cisco Systems, Inc.
                                                              E. Marocco
                                                          Telecom Italia
                                                      September 29, 2013


  CUSAX: Combined Use of the Session Initiation Protocol (SIP) and the
           Extensible Messaging and Presence Protocol (XMPP)
                        draft-ivov-xmpp-cusax-08

Abstract

   This document suggests some strategies for the combined use of the
   Session Initiation Protocol (SIP) and the Extensible Messaging and
   Presence Protocol (XMPP).  Such strategies 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 document does
   not define any new protocols or syntax for either SIP or XMPP, and by
   intent it does not attempt to standardize "best current practices".
   However, implementing the suggested strategies outlined in this
   document 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 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."




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   This Internet-Draft will expire on April 02, 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
   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  . . . . . . . . . . . . . . . . . . . . . .   5
   3.  Operation . . . . . . . . . . . . . . . . . . . . . . . . . .   7
     3.1.  Server-Side Setup . . . . . . . . . . . . . . . . . . . .   7
     3.2.  Service Management  . . . . . . . . . . . . . . . . . . .   7
     3.3.  Client-Side Discovery and Usability . . . . . . . . . . .   8
     3.4.  Indicating a Relation Between SIP and XMPP Accounts . . .   9
     3.5.  Matching Incoming SIP Calls to XMPP JIDs  . . . . . . . .  10
   4.  Multi-Party Interactions  . . . . . . . . . . . . . . . . . .  11
   5.  Federation  . . . . . . . . . . . . . . . . . . . . . . . . .  12
   6.  Summary of Suggested Strategies . . . . . . . . . . . . . . .  13
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  14
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  15
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  15
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  16
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  16
   Appendix A.  Acknowledgements . . . . . . . . . . . . . . . . . .  18
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  18

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
   [XEP-0045].




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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 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 elaborated features such as
      multi-party chat, server-stored contact lists, and file transfer
      [RFC6914].

   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 [XEP-0166] and arguably their most
      popular use case is audio and video calling [XEP-0167].

   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 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

   This document explains how such hybrid offerings can be achieved with
   a minimum of modifications to existing software while providing an



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   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 it also suggests 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 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
   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).  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 or obtained from a user directory).  However, the rules to
   follow in such cases are left to application developers (although
   they might be described in a future document).













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   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
   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
   [RFC2026].  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.

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
   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-
   optimal.

   While it should be possible for CUSAX users to manually configure
   their separate SIP and XMPP accounts (often by means of so-called
   "wizard" interfaces), service providers offering CUSAX services to
   users of dual-stack SIP/XMPP clients ought to provide means of online
   provisioning, typically by means of a web-based service at an HTTP
   URI (naturally single-purpose SIP services or XMPP services could
   offer online provisioning as well, but they can be especially helpful
   where the two aspects of the CUSAX service need to have several



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   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 adopted; however, application
   developers are encouraged to explore the potential for future
   progress in that space (e.g., perhaps based on technologies such as
   WebFinger [I-D.ietf-appsawg-webfinger]).

   A CUSAX client should make it possible for a user to configure the
   software so that a given feature (say, video conferencing or textual
   chat) to use by default a given protocol (say, SIP for video
   conferencing and XMPP for textual chat).  In particular, it is
   suggested that CUSAX clients allow for audio and video calling
   features to be disabled over XMPP, and for instant messaging and
   presence features to be disabled over SIP.  It is a matter of
   implementation whether a CUSAX client enables a user to override such
   defaults in various ways (e.g., by domain, by individual contact, or
   by device).

   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.

   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
   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 its contacts but calling it



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   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, would
   typically be disabled in the XMPP stack, so any form of 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
   allow CUSAX clients to fall back and seamlessly use SIP for these
   features.

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.  (The same considerations apply even if the
   directory storage format is not vCard.)

   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 validating the values of vCard "tel" fields
   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.

3.2.  Service Management





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   The task of operating and managing a SIP service or an 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.

   o  The potential for subtle differences in security posture across
      SIP and XMPP (e.g., SIP servers and XMPP servers might support
      different 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, 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 standalone SIP or XMPP services.
   Providers of CUSAX services are advised to consider the potential for
   such topologies to cause operational challenges.

3.3.  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 or stored in directories in order to be dialed and called.



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   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 [RFC6350] as the basis for choosing
   whether to enable or disable the possibility for starting video calls
   (i.e., if there is no "video" tel type for a particular contact, do
   not provide a way for the user to start a video call with 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 [XEP-0152].  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 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.

3.4.  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 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.



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   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 relation 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 vCard was
   obtained or a SIP account with another provider).

   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 complete audio and video calls
   with XMPP users who are not part of a CUSAX deployment.

3.5.  Matching Incoming SIP Calls to XMPP JIDs

   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 [RFC6993].  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: <xmpp:alice@xmpp.example.com> ;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.





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   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.

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
   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.  These methods would enable a CUSAX-aware SIP
   conference server to advertise the existence of an associated XMPP
   chatroom, and for a CUSAX-aware XMPP chatroom to advertise the
   existence of an associated SIP conference server.

   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.

   At the level of the CUSAX service, it is also possible to enforce
   tighter integration between the XMPP MUC and the SIP conference.
   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



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   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 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 in conference calling.

5.  Federation

   In theory there are no technical reasons why federation 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 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 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 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.








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   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 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 inter-domain numbers are flagged with
   an appropriate audio or textual warning.

6.  Summary of Suggested Strategies

   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) in
        which case use SIP.

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

   4.   Provide on-line 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.

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



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   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" acccount.

   10.  Optionally, determine the identity of the audio/video caller and
        a corresponding XMPP roster entry so that the user could revert
        to 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 setup two separate accounts for your 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 or a phone number that
       are reachable through your 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 your SIP INVITE messages, so that clients can more rapidly
       associate a caller with a roster entry and display a "Caller ID".

7.  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.  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 [RFC3261] and SCRAM authentication [RFC5802]



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      for XMPP as specified in [RFC6120]).  Because SIP uses a password-
      based method (digest) and XMPP uses a pluggable framework for
      authentication via the Simple Authentication and Security Layer
      (SASL) technology [RFC4422], 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) [RFC5246] 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.

   Section 5 discusses potential issues that may arise due to a mismatch
   between client capabilities, such as calls being initiated with costs
   contrary to the expectation of 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 Section 5.

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

8.  IANA Considerations

   This document has no actions for the IANA.

9.  References





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9.1.  Normative References

   [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.

   [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.

9.2.  Informative References

   [I-D.ietf-appsawg-webfinger]
              Jones, P., Salgueiro, G., Jones, M., and J. Smarr,
              "WebFinger", draft-ietf-appsawg-webfinger-18 (work in
              progress), August 2013.

   [I-D.ivov-grouptextchat-purpose]
              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-03 (work in progress), June 2013.

   [RFC2026]  Bradner, S., "The Internet Standards Process -- Revision
              3", BCP 9, RFC 2026, October 1996.

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353, February
              2006.

   [RFC4422]  Melnikov, A. and K. Zeilenga, "Simple Authentication and
              Security Layer (SASL)", RFC 4422, June 2006.

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

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

   [RFC6914]  Rosenberg, J., "SIMPLE Made Simple: An Overview of the
              IETF Specifications for Instant Messaging and Presence
              Using the Session Initiation Protocol (SIP)", RFC 6914,
              April 2013.

   [RFC6993]  Saint-Andre, P., "Instant Messaging and Presence Purpose
              for the Call-Info Header Field in the Session Initiation
              Protocol (SIP)", RFC 6993, July 2013.

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

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

   [XEP-0045]
              Saint-Andre, P., "Multi-User Chat", XSF XEP 0045, February
              2012.

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

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

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

   [XEP-0166]
              Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
              S., and J. Hildebrand, "Jingle", XSF XEP 0166, December
              2009.

   [XEP-0167]
              Ludwig, S., Saint-Andre, P., Egan, S., McQueen, R., and D.
              Cionoiu, "Jingle RTP Sessions", XSF XEP 0167, December
              2009.

   [XEP-0292]
              Saint-Andre, P. and S. Mizzi, "vCard4 Over XMPP", XSF XEP
              0292, September 2013.



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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 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, 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.

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









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

   Email: enrico.marocco@telecomitalia.it












































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