Network Working Group E. Ivov
Internet-Draft Jitsi
Intended status: Informational P. Saint-Andre
Expires: October 06, 2013 Cisco Systems, Inc.
E. Marocco
Telecom Italia
April 04, 2013
CUSAX: Combined Use of the Session Initiation Protocol (SIP) and the
Extensible Messaging and Presence Protocol (XMPP)
draft-ivov-xmpp-cusax-04
Abstract
This document describes suggested 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 (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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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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material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 06, 2013.
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Copyright Notice
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document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Client Bootstrap . . . . . . . . . . . . . . . . . . . . . . 4
3. Operation . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Multi-Party Interactions . . . . . . . . . . . . . . . . . . 8
5. Federation . . . . . . . . . . . . . . . . . . . . . . . . . 8
6. Security Considerations . . . . . . . . . . . . . . . . . . . 9
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
8. Informative References . . . . . . . . . . . . . . . . . . . 9
Appendix A. Acknowledgements . . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11
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].
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.
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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 arguably their most popular use
case is audio and video calling.
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-based software. XMPP implementations, on the
other hand, greatly outnumber and outperform those available for
instant messaging and presence extensions developed 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.
+------------+ +-------------+
| 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
optimal user experience. It tries to cover points such as server
discovery, determining a SIP 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 recommends
certain server-side configurations.
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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.
Finally, this document concentrates on use cases where the SIP
services and XMPP services are controlled by one and the same
provider. Since this document is of an informational nature, it is
not unreasonable for clients to apply some of the guidelines here
even in cases where there is no established relationship between the
SIP and the XMPP services (for example, it is reasonable for a client
to provide a way for its users to easily start a call to a phone
number recorded in a vCard). However, the exact set of rules to
follow in such cases is left to application developers.
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, making it rather
complicated for inexperienced users to configure a mail client and
start using it with a new service. As a result, Internet service
providers often need to provide configuration instructions for
various mail clients. Client developers and communication device
manufacturers on the other hand often ship with a number of 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, 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 a service provider 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
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make it possible for such features to be disabled for a specific
account. In particular, it is suggested that clients allow for audio
and video calling features to be disabled for XMPP accounts, and that
instant messaging and presence features should also be made optional
for SIP accounts.
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 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 XMPP CUSAX 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 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.
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 [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.
Additionally, clients that have separete triggers (buttons) for audio
and video calls may choose to use the presence or absence of the
"video" tel type defined in [RFC6350] and enable or disable the
possibility for starting video calls accordingly.
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
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validation 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.
+--------------+
| Provisioning |-----------+
| Server | |
+--------------+ v
| +----------------+
| | vCard Storage/ |
| | User Directory |
| +----------------+
| / \
| +------------+ +-------------+
| | SIP Server | | XMPP Server |
| +------------+ +-------------+
| \ /
| media \ / instant messaging,
| signaling \ / presence, etc.
| \ /
| +--------------+
+---------------| CUSAX Client |
+--------------+
Figure 2: Example Deployment
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 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 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 order to provide a similar experience when the user has multiple
telephony-capable accounts, client implementers may choose to
indicate the existence of a special relationship between the SIP and
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XMPP accounts of a CUSAX service. For example, let's say that
Alice's service provider has opened both an XMPP account and a SIP
account for her. During or after provisioning, her client could
indicate that alice@xmpp.example.com has a CUSAX 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 between
the SIP and XMPP accounts of a CUSAX service and the client is aware
of multiple telephony-capable accounts, it ought to present the user
with the choice of reaching the contact through any of those
accounts. This includes the source XMPP account where the vCard was
obtained (in case its telephony capabilities are not disabled through
configuration or provisioning), 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 for alternative communication methods as advertised in
XMPP presence broadcasts and 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 audio/video features were
disabled in the XMPP stack.
When receiving SIP calls, clients may wish to determine the identity
of the caller and a corresponding XMPP 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.
In addition, in order to avoid the effort of iterating over an entire
roster and retrieving all vCards, CUSAX clients may use a SIP Call-
Info header whose 'purpose' token field parameter has a value of
"impp" as described in [I-D.saintandre-impp-call-info] such as the
following:
Call-Info: <xmpp:alice@example.com> ;purpose=impp
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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 a vCard entry. If this confirmation succeeds the client
would not need to search the entire roster and retrieve all vCards.
Not performing the check would allow any caller (including malicious
ones) to employ someone else's identity and perform various scams or
Man-in-the-Middle attacks.
4. Multi-Party Interactions
This document concentrates on problems related to one-to-one
communication. While it is possible for clients and other
specifications to build upon this and provide suggestions for
improving the Unified Communications user experience in cases of
multi-user chats in conference calling (e.g., ways of mapping XMPP
Multi-User Chatrooms to conference calls and vice versa), such
mechanisms are considered out of scope for this version of CUSAX.
5. Federation
In theory there are no technical reasons why federation would require
special behaviour 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 specification. 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
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.
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This could be a problem because, other than the fact that Bob's
address is from a different domain, Alice would have no obvious and
straightforward cues telling her that this is in fact a call to the
PSTN. In addition to the potentially lower audio quality, Alice may
also end up incurring unexpected charges for such calls.
In order to avoid such issues, providers maintaining a CUSAX service
for the users in their domain may choose to provide additional cues
(e.g., a user interface warning or an an audio tone or 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 extra-domain numbers are flagged with
an appropriate audio or textual warning.
6. 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. For
example, the SIP aspect and XMPP aspect of the CUSAX service might
offer different authentication options (e.g., digest authentication
for SIP as specified in [RFC3261] and SCRAM authentication [RFC5802]
for XMPP as specified in [RFC6120]). Similarly, a CUSAX client might
successfully negotiate Transport Layer Security (TLS) [RFC5246] when
connecting to the XMPP aspect of the service but not when connecting
to the SIP aspect. Such mismatches could introduce the possibility
of downgrade attacks. User agent developers and service providers
ought to ensure that such mismatches are avoided as much as possible.
Refer to the specifications for the relevant SIP and XMPP features
for detailed security considerations applying to each "stack" in a
CUSAX client.
7. IANA Considerations
This document has no actions for the IANA.
8. Informative References
[I-D.saintandre-impp-call-info]
Saint-Andre, P., "Instant Messaging and Presence Purpose
for the Call-Info Header in the Session Initiation
Protocol (SIP)", draft-saintandre-impp-call-info-00 (work
in progress), March 2013.
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[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.
[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.
[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.
[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.
[RFC6350] Perreault, S., "vCard Format Specification", RFC 6350,
August 2011.
[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-0152]
Hildebrand, J. and P. Saint-Andre, "XEP-0152: Reachability
Addresses", XEP XEP-0152, February 2013.
[XEP-0292]
Saint-Andre, P. and S. Mizzi, "vCard4 Over XMPP", XSF XEP
0292, October 2011.
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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 persons for their
reviews and suggestions: Aaron M. Evans, Sebastien Couture, Olivier
Crete, Kevin Gallagher, Adrian Georgescu, Saul Ibarra Corretge,
Daniel Pocock, Travis Reitterd, and Gonzalo Salgueiro.
Authors' Addresses
Emil Ivov
Jitsi
Strasbourg 67000
France
Phone: +33-672-811-555
Email: emcho@jitsi.org
Peter Saint-Andre
Cisco Systems, Inc.
1899 Wynkoop Street, Suite 600
Denver, CO 80202
USA
Phone: +1-303-308-3282
Email: psaintan@cisco.com
Enrico Marocco
Telecom Italia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: enrico.marocco@telecomitalia.it
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