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Network Working Group                                            E. Ivov
Internet-Draft                                                     Jitsi
Intended status: BCP                                     E. Marocco, Ed.
Expires: September 6, 2012                                Telecom Italia
                                                           March 5, 2012


     Combined Use of the Session Initiation Protocol (SIP) and the
           eXtensible Messaging and Presence Protocol (CUSAX)
                        draft-ivov-xmpp-cusax-00

Abstract

   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 complimenting
   subsets of features from each of the protocols.  Typically such
   subsets would include telephony oriented from SIP and instant
   messaging and presence capabilities from XMPP.  This specification
   does not define any new protocols or syntax for neither SIP nor 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 us 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
   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
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on September 6, 2012.

Copyright Notice




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   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
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
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   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.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 4
   3.  Client Bootstrap  . . . . . . . . . . . . . . . . . . . . . . . 4
   4.  Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
   5.  Security Considerations . . . . . . . . . . . . . . . . . . . . 6
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 6
   7.  References  . . . . . . . . . . . . . . . . . . . . . . . . . . 6
     7.1.  Normative References  . . . . . . . . . . . . . . . . . . . 6
     7.2.  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 its instant messaging and presence
   capabilities.

   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.

   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.

   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.

   Yet, 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 protocols
   recommended by SIMPLE, such as [MSRP] and [XCAP].

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

   Most often such combined use would employ SIP exclusively for audio,
   video and telephony services and it would rely on XMPP for anything
   else varying from chat, roster management and presence to exchanging
   files.

   This document explains how the above could be achieved with a minimum
   amount of modifications on existing software while providing an



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   optimal user experience.  It tries to cover points such as server
   discovery, determining a SIP AOR while using XMPP and an XMPP JID
   from incoming SIP requests.  Most of the text here pertains to client
   behavior but it also recommends certain server-side configurations.


2.  Terminology

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


3.  Client Bootstrap

   One of the main problems of using two distinct protocols when
   providing one service is how it affects usability.  E-mail services
   for example have long been affected by the mixed use of SMTP on for
   outgoing mail and POP3 and IMAP for incoming, 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 allow to
   easily set up a new account for a number of popular e-mail services.
   While this may improve the situation to some extent, user experience
   is still clearly sub-optimal.

   While it should be possible for CUSAX users to manually configure
   their separate SIP and XMPP accounts, it is RECOMMENDED that dual
   stack SIP/XMPP clients provide means of online provisioning.  While
   the specifics of such mechanisms are not in 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. user
   id and password).

   Given that many of the features that CUSAX 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.
   Specifically it is RECOMMENDED that clients allow for audio/video
   calling features to be disabled for XMPP accounts.  Additionally
   instant messaging and presence features MAY 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 SHOULD log independently
   to the SIP and XMPP accounts that make up the CUSAX service and
   should 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 implementors.  The point is
   that such presentation could help users better understand why they
   are being shown two different connections for a single service.  It
   could even alleviate especially situations where one of these
   connections is disrupted while the other one is successfully
   maintained.


4.  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] "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).  It SHOULD however be reachable through the SIP
   counterpart of this CUSAX service.

   In order to make sure that the above is always respected, service
   maintainers MAY prevent clients (and hence users) from modifying the
   VCARD "tel" fields or they MAY apply some form of validation before
   recording changes.

   When rendering the XMPP roaster 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 SHOULD do so immediately upon user request without
   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 SHOULD present the user with the choice of reaching the



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   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.

   The client should use XMPP for all other forms of communication with
   the contacts from its roster so it should and this should occur
   naturally given that they were retrieved through XMPP.

   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 their XMPP JID, but at
   this point we are not really sure if that's ' such a good idea.
   (After all Contact headers carry URIs and JIDs are not URIs).


5.  Security Considerations

   TBD


6.  Acknowledgements

   This draft is inspired by work from Markus Isomaki and Simo
   Veikkolainen.


7.  References

7.1.  Normative References

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

7.2.  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.

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



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   [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.

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

   [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.



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   [XEP-0177]
              Beda, J., Saint-Andre, P., Hildebrand, J., and S. Egan,
              "XEP-0177: Jingle Raw UDP Transport Method", XEP XEP-0177,
              December 2009.


Authors' Addresses

   Emil Ivov
   Jitsi
   Strasbourg  67000
   France

   Email: emcho@jitsi.org


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

   Email: enrico.marocco@telecomitalia.it




























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