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draft-ivov-xmpp-cusax-01.txt
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draft-ivov-xmpp-cusax-01.txt
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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)
draft-ivov-xmpp-cusax-01
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 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
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."
This Internet-Draft will expire on December 7, 2012.
Ivov, et al. Expires December 7, 2012 [Page 1]
Internet-Draft Combined Use of SIP and XMPP June 2012
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
(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 . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Client Bootstrap . . . . . . . . . . . . . . . . . . . . . . . 4
3. Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Security Considerations . . . . . . . . . . . . . . . . . . . . 6
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6
6. Informative References . . . . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 8
Ivov, et al. Expires December 7, 2012 [Page 2]
Internet-Draft Combined Use of SIP and XMPP June 2012
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
Ivov, et al. Expires December 7, 2012 [Page 3]
Internet-Draft Combined Use of SIP and XMPP June 2012
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
password).
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.
Ivov, et al. Expires December 7, 2012 [Page 4]
Internet-Draft Combined Use of SIP and XMPP June 2012
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
maintained.
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
Ivov, et al. Expires December 7, 2012 [Page 5]
Internet-Draft Combined Use of SIP and XMPP June 2012
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
[XEP-0152].
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
TBD
5. Acknowledgements
This draft is inspired by work from Markus Isomaki and Simo
Veikkolainen.
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.
Ivov, et al. Expires December 7, 2012 [Page 6]
Internet-Draft Combined Use of SIP and XMPP June 2012
[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.
Ivov, et al. Expires December 7, 2012 [Page 7]
Internet-Draft Combined Use of SIP and XMPP June 2012
[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.
[XEP-0152]
Hildebrand, J. and P. Saint-Andre, "XEP-0152: Reachability
Addresses", XEP XEP-0152, October 2008.
Authors' Addresses
Emil Ivov
Jitsi
Strasbourg 67000
France
Phone: +33-672-811-555
Email: [email protected]
Enrico Marocco
Telecom Italia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: [email protected]
Ivov, et al. Expires December 7, 2012 [Page 8]
Internet-Draft Combined Use of SIP and XMPP June 2012
Peter Saint-Andre
Cisco Systems, Inc.
1899 Wynkoop Street, Suite 600
Denver, CO 80202
USA
Phone: +1-303-308-3282
Email: [email protected]
Ivov, et al. Expires December 7, 2012 [Page 9]