Network Working Group                                            E. Ivov
Internet-Draft                                                     Jitsi
Intended status: Informational                                E. Marocco
Expires: December 7, 2012                                 Telecom Italia
                                                          P. Saint-Andre
                                                     Cisco Systems, Inc.
                                                            June 5, 2012

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


   This document describes current practices for combined use of the
   Session Initiation Protocol (SIP) and the Extensible Messaging and
   Presence Protocol (XMPP).  Such practices aim to provide a single
   fully featured real-time communication service by using complementary
   subsets of features from each of the protocols.  Typically such
   subsets would include telephony capabilities from SIP and instant
   messaging and presence capabilities from XMPP.  This specification
   does not define any new protocols or syntax for either SIP or XMPP.
   However, implementing it may require modifying or at least
   reconfiguring existing client and server-side software.  Also, it is
   not the purpose of this document to make recommendations as to
   whether or not such combined use should be preferred to the
   mechanisms provided natively by each protocol like 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
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   Internet-Drafts are draft documents valid for a maximum of six months
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   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on December 7, 2012.

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Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . . . 3
   2.  Client Bootstrap  . . . . . . . . . . . . . . . . . . . . . . . 4
   3.  Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
   4.  Security Considerations . . . . . . . . . . . . . . . . . . . . 6
   5.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 6
   6.  Informative References  . . . . . . . . . . . . . . . . . . . . 6
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . . 8

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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.

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

   o  Today, in the context of the SIMPLE working group, the IETF has
      defined a number of protocols and protocol extensions that not
      only allow for SIP to be used for regular instant messaging and
      presence but that also provide mechanisms for elaborated features
      such as multi-user chats, server-stored contact lists, file
      transfer and others.
   o  Similarly, the XMPP community and the XMPP Standards Foundation
      have worked on defining a number of XMPP Extension Protocols
      (XEPs) that provide XMPP implementations with the means of
      establishing end-to-end sessions.  These extensions are often
      jointly referred to as Jingle and their arguably most popular use
      case are audio and video calls.

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

   For these reasons, in a number of cases adopters have found
   themselves needing a set of features that are not offered by any
   single-protocol solution but that separately exist in SIP and XMPP
   products.  The idea of seamlessly using both protocols together would
   hence often appeal to service providers.

   Most often the combined use of SIP and XMPP ("CUSAX") 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.

   This document explains how such hybrid offerings can be achieved with
   a minimum of modifications to existing software while providing an
   optimal user experience.  It tries to cover points such as server

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   discovery, determining a SIP AOR while using XMPP and determining an
   XMPP JabberID from incoming SIP requests.  Most of the text here
   pertains to client behavior but it also recommends certain server-
   side configurations.

   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 to
   an XMPP network.  A separate series of documents has been produced
   that defines such mappings.

2.  Client Bootstrap

   One of the main problems of using two distinct protocols when
   providing one service is the effect on usability.  E-mail services,
   for example, have long been affected by the mixed use of SMTP on for
   outgoing mail and POP3 or IMAP for incoming mail, making it rather
   complicated for inexperienced users to configure a mail client and
   start using it with a new service.  As a result, mailing list
   services often need to provide configuration instructions for various
   mail clients.  Client developers and communications 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 e-mail 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, dual-stack SIP/XMPP clients
   ought to provide means of online provisioning.  While the specifics
   of such mechanisms are outside the scope of this specification, they
   should make it possible for service providers to remotely configure
   the clients based on minimal user input (e.g., only a user ID and

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

   The main advantage of the above would be that clients would be able
   to continue to function properly and use the complete feature set of
   stand-alone SIP and XMPP accounts.

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   Once client bootstrap has completed, clients need to log in
   independently to the SIP and XMPP accounts that make up the CUSAX
   "service" and then maintain both these connections.  In order to
   improve user experience, when reporting connection status clients may
   also wish to present the CUSAX XMPP connection as an "instant
   messaging" or a "chat" account.  Similarly they could also depict the
   SIP CUSAX 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 provided by a single entity, 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 successfully

3.  Operation

   Once a CUSAX client has been provisioned/configured to connect to the
   corresponding SIP and XMPP services it would proceed by retrieving
   its XMPP roster.  In order for CUSAX to function properly, XMPP
   service administrators should make sure that at least one of the
   VCARD [RFC4825] "tel" fields for each contact is properly populated
   with a SIP URI or a phone number.  There are no limitations as to the
   form of that number (e.g. it does not need to respect any equivalence
   with the XMPP JID).  However, it ought to be reachable through the
   SIP counterpart of this CUSAX service.

   To ensure that the foregoing approach is always respected, service
   providers might consider (1) preventing clients (and hence users)
   from modifying the VCARD "tel" fields or (2) applying some form of
   validation before recording changes.  Of course such validation would
   be feasible mostly in cases where one single provider controls both
   the XMPP and the SIP service since such providers would "know" what
   SIP AOR corresponds to a given XMPP user.

   When rendering the XMPP roster CUSAX clients should make sure that
   users are presented with a "Call" option for each roster entry that
   has a properly set "tel" field even if calling has been disabled for
   that particular XMPP account.  The usefulness of such a feature is
   not limited to CUSAX.  After all, numbers are entered in VCARDs in
   order to be dialed and called.  Hence, as long as an XMPP client is
   equipped with accounts that have calling features it may wish to
   present the user with the option of using these accounts to reach
   numbers from an XMPP VCARD.  In order to improve usability, in cases
   where clients are provisioned with only a single telephony-capable
   account they ought to do so immediately upon user request without

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   asking for confirmation.  This way CUSAX users whose only account
   with calling capabilities would often be the SIP part of their
   service, would be having better user experience.  If on the other
   hand, the CUSAX client is aware of multiple telephony-capable
   accounts, it ought to present the user with the choice of reaching
   the phone number through any of them (including the source XMPP
   account where the VCARD was obtained) in order to guarantee proper
   operation for XMPP accounts that are not part of a CUSAX deployment.

   In addition to discovering phone numbers from VCARDs, clients may
   also check presence broadcasts and the appropriate Personal Eventing
   Protocol nodes as described in XEP-0152: Reachability Addresses

   The client should use XMPP for all other forms of communication with
   the contacts from its roster, which will occur naturally because they
   were retrieved through XMPP and only voice/video features were
   disabled in the XMPP stack.

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

   An alternate mechanism would be for CUSAX clients to add to their SIP
   invite requests a Contact header containing the XMPP URI
   corresponding to their JID as per [RFC4622].

4.  Security Considerations


5.  Acknowledgements

   This draft is inspired by work from Markus Isomaki and Simo

6.  Informative 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.

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   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4622]  Saint-Andre, P., "Internationalized Resource Identifiers
              (IRIs) and Uniform Resource Identifiers (URIs) for the
              Extensible Messaging and Presence Protocol (XMPP)",
              RFC 4622, July 2006.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

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

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

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
              Mail Extensions (S/MIME) Version 3.2 Message
              Specification", RFC 5751, January 2010.

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   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC5853]  Hautakorpi, J., Camarillo, G., Penfield, R., Hawrylyshen,
              A., and M. Bhatia, "Requirements from Session Initiation
              Protocol (SIP) Session Border Control (SBC) Deployments",
              RFC 5853, April 2010.

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

   [RFC6189]  Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Unicast Secure RTP", RFC 6189,
              April 2011.

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

              Hildebrand, J. and P. Saint-Andre, "XEP-0152: Reachability
              Addresses", XEP XEP-0152, October 2008.

Authors' Addresses

   Emil Ivov
   Strasbourg  67000

   Phone: +33-672-811-555

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


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

   Phone: +1-303-308-3282

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