Network Working Group E. Ivov
Internet-Draft Jitsi
Intended status: Informational P. Saint-Andre
Expires: December 08, 2013 Cisco Systems, Inc.
E. Marocco
Telecom Italia
June 06, 2013
CUSAX: Combined Use of the Session Initiation Protocol (SIP) and the
Extensible Messaging and Presence Protocol (XMPP)
draft-ivov-xmpp-cusax-06
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 the practices 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
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 08, 2013.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Client Bootstrap . . . . . . . . . . . . . . . . . . . . . . 4
3. Operation . . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Server-Side Setup . . . . . . . . . . . . . . . . . . . . 7
3.2. Client-Side Discovery and Usability . . . . . . . . . . . 7
3.3. Indicating a Relation Between SIP and XMPP Accounts . . . 8
3.4. Matching Incoming SIP Calls to XMPP JIDs . . . . . . . . 9
4. Multi-Party Interactions . . . . . . . . . . . . . . . . . . 9
5. Federation . . . . . . . . . . . . . . . . . . . . . . . . . 10
6. Summary of Suggested Practices . . . . . . . . . . . . . . . 11
7. Security Considerations . . . . . . . . . . . . . . . . . . . 13
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
9. Informative References . . . . . . . . . . . . . . . . . . . 14
Appendix A. Acknowledgements . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15
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:
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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 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].
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. 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
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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 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 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 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 exact set of
rules to follow in such cases is left to application developers.
Finally, 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).
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
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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, 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 configuration options in common). 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 prefer 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 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 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 as displayed in Figure 2.
+--------------+
| Provisioning |-----------+
| Server | |
+--------------+ v
| +----------------+
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| | User Directory |
| +----------------+
| / \
| +------------+ +-------------+
| | SIP Server | | XMPP Server |
| +------------+ +-------------+
| \ /
| media \ / instant messaging,
| signaling \ / presence, etc.
| \ /
| +--------------+
+---------------| CUSAX Client |
+--------------+
Figure 2: Example Deployment
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
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, are
disabled in the XMPP stack, so any form of communication based on
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these features (e.g. direct calls, conferences, desktop streaming,
etc.) will 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 (1) preventing clients (and hence users)
from modifying the vCard "tel" fields or (2) applying some form of
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 (as indicated in Figure 2).
3.2. 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.
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
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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.
3.3. 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.
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
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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 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.
3.4. Matching Incoming SIP Calls to XMPP JIDs
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' header field parameter has a value of
"impp" as described in [I-D.saintandre-impp-call-info]. 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 a vCard entry 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.
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
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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.
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.
Of course, tighter integration between the XMPP MUC and the SIP
conference is also possible. 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 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 specification, this approach is by no means normative.
Implementation and future specifications 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
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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 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
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.
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 inter-domain numbers are flagged with
an appropriate audio or textual warning.
6. Summary of Suggested Practices
The following practices are suggested for CUSAX user agents:
1. By default, prefer SIP for audio and video, and XMPP for
messaging and presence.
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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.
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 practices are suggested for CUSAX services:
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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. 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.
8. IANA Considerations
This document has no actions for the IANA.
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9. Informative References
[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-01 (work in progress), May 2013.
[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-04 (work
in progress), May 2013.
[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.
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353, February
2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[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.
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[RFC6350] Perreault, S., "vCard Format Specification", RFC 6350,
August 2011.
[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-0292]
Saint-Andre, P. and S. Mizzi, "vCard4 Over XMPP", XSF XEP
0292, October 2011.
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.
Authors' Addresses
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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
Enrico Marocco
Telecom Italia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: enrico.marocco@telecomitalia.it
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