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Internet Engineering Task Force (IETF) A. Johnston
Request for Comments: 6567 Avaya
Category: Informational L. Liess
ISSN: 2070-1721 Deutsche Telekom AG
April 2012
Problem Statement and Requirements for
Transporting User-to-User Call Control Information in SIP
Abstract
This document introduces the transport of call control User-to-User
Information (UUI) using the Session Initiation Protocol (SIP) and
develops several requirements for a new SIP mechanism. Some SIP
sessions are established by or related to a non-SIP application.
This application may have information that needs to be transported
between the SIP User Agents during session establishment. In
addition to interworking with the Integrated Services Digital Network
(ISDN) UUI Service, this extension will also be used for native SIP
endpoints requiring application UUI.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc6567.
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
Johnston & Liess Informational [Page 1]
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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. Overview ........................................................2
2. Use Cases .......................................................3
2.1. User Agent to User Agent ...................................3
2.2. Proxy Retargeting ..........................................4
2.3. Redirection ................................................4
2.4. Referral ...................................................5
3. Requirements ....................................................6
4. Security Considerations .........................................8
5. Acknowledgements ...............................................10
6. Informative References .........................................10
1. Overview
This document describes the transport of User-to-User Information
(UUI) during SIP [RFC3261] session setup. This section introduces
UUI and explains how it relates to SIP.
We define SIP UUI data as application-specific information that is
related to a session being established using SIP. It is assumed that
the application is running in both endpoints in a two-party session.
That is, the application interacts with both the User Agents in a SIP
session. In order to function properly, the application needs a
small piece of information, the UUI, to be transported at the time of
session establishment. This information is essentially opaque data
to SIP -- it is unrelated to SIP routing, authentication, or any
other SIP function. This application can be considered to be
operating at a higher layer on the protocol stack. As a result, SIP
should not interpret, understand, or perform any operations on the
UUI. Should this not be the case, then the information being
transported is not considered UUI, and another SIP-specific mechanism
will be needed to transport the information (such as a new header
field). In particular, this mechanism creates no requirements on
intermediaries such as proxies, Back-to-Back User Agents, and Session
Border Controllers.
UUI is defined this way for two reasons. First, this definition
supports a strict layering of protocols and data. Providing
information and understanding of the UUI to the transport layer (SIP
in this case) would not provide any benefits and instead could create
cross-layer coupling. Second, it is neither feasible nor desirable
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for a SIP User Agent (UA) to understand the information; instead, the
goal is for the UA to simply pass the information as efficiently as
possible to the application that does understand the information.
An important application is the interworking with User-to-User
Information (UUI) in ISDN, specifically the transport of the call-
control-related ITU-T Q.931 User-to-User Information Element (UUIE)
[Q931] and ITU-T Q.763 User-to-User Information Parameter [Q763] data
in SIP. ISDN UUI is widely used in the Public Switched Telephone
Network (PSTN) today in contact centers and call centers. These
applications are currently transitioning away from using ISDN for
session establishment to using SIP. Native SIP endpoints will need
to implement a similar service and be able to interwork with this
ISDN service.
Note that the distinction between call control UUI and non-call-
control UUI is very important. SIP already has a mechanism for
sending arbitrary UUI data between UAs during a session or dialog --
the SIP INFO [RFC6086] method. Call control UUI, in contrast, must
be exchanged at the time of setup and needs to be carried in the
INVITE and a few other methods and responses. Applications that
exchange UUI but do not have a requirement that it be transported and
processed during call setup can simply use SIP INFO and do not need a
new SIP extension.
In this document, four different use case call flows are discussed.
Next, the requirements for call control UUI transport are discussed.
2. Use Cases
This section discusses four use cases for the transport of call
control User-to-User Information. These use cases will help motivate
the requirements for SIP call control UUI.
2.1. User Agent to User Agent
In this scenario, the originating UA includes UUI in the INVITE sent
through a proxy to the terminating UA. The terminating UA can use
the UUI in any way. If it is an ISDN gateway, it could map the UUI
into the appropriate DSS1 [Q933] information element, QSIG [QSIG]
information element, or ISDN User Part (ISUP) parameter.
Alternatively, the using application might render the information to
the user or use it during alerting or as a lookup for a screen pop.
In this case, the proxy does not need to understand the UUI
mechanism, but normal proxy rules should result in the UUI being
forwarded without modification. This call flow is shown in Figure 1.
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Originating UA Proxy Terminating UA
| | |
| INVITE (UUI) F1 | |
|------------------->| INVITE (UUI) F2 |
| 100 Trying F3 |------------------->|
|<-------------------| 200 OK F4 |
| 200 OK F5 |<-------------------|
|<-------------------| |
| ACK F6 | |
|------------------->| ACK F7 |
| |------------------->|
Figure 1: Call Flow with UUI Exchanged between Originating and
Terminating UAs
2.2. Proxy Retargeting
In this scenario, the originating UA includes UUI in the INVITE
request sent through a proxy to the terminating UA. The proxy
retargets the INVITE request, changing its Request-URI to a URI that
addresses the terminating UA. The UUI data is then received and
processed by the terminating UA. This call flow is identical to
Figure 1 except that the proxy retargets the request, i.e., changes
the Request-URI as directed by some unspecified process. The UUI in
the INVITE request needs to be passed unchanged through this proxy
retargeting operation. Note that the contents of the UUI is not used
by the proxy for routing, as the UUI has only end-to-end significance
between UAs.
2.3. Redirection
In this scenario, UUI is inserted by an application that utilizes a
SIP Redirect Server. The UUI is then included in the INVITE request
sent by the originating UA to the terminating UA. In this case, the
originating UA does not necessarily need to support the UUI mechanism
but does need to support the SIP redirection mechanism used to
include the UUI data. Two examples of UUI with redirection (transfer
and diversion) are defined in [ANSI] and [ETSI].
Note that this case may not precisely map to an equivalent ISDN
service use case. This is because there is no one-to-one mapping
between elements in a SIP network and elements in an ISDN network.
Also, there is not an exact one-to-one mapping between SIP call
control and ISDN call control. However, this should not prevent the
usage of SIP call control UUI in these cases. Instead, these slight
differences between the SIP UUI mechanism and the ISDN service need
to be carefully noted and discussed in an interworking specification.
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Figure 2 shows this scenario, with the Redirect Server inserting UUI
that is then included in the INVITE request F4 sent to the
terminating UA.
Originating UA Redirect Server Terminating UA
| | |
| INVITE F1 | |
|------------------->| |
| 302 Moved (UUI) F2 | |
|<-------------------| |
| ACK F3 | |
|------------------->| |
| INVITE (UUI) F4 | |
|---------------------------------------->|
| 200 OK F5 |
|<----------------------------------------|
| ACK F6 |
|---------------------------------------->|
Figure 2: Call Flow with UUI Exchanged between Redirect Server and
Terminating UA
A common example application of this call flow is an Automatic Call
Distributer (ACD) in a PSTN contact center. The originator would be
a PSTN gateway. The ACD would act as a Redirect Server, inserting
UUI based on called number, calling number, time of day, and other
information. The resulting UUI would be passed to the agent's
handset which acts as the terminating UA. The UUI could be used to
lookup information for rendering to the agent at the time of call
answering.
This redirection scenario and the referral scenario in the next
section are the most important scenarios for contact center
applications. Incoming calls to a contact center almost always are
redirected or referred to a final destination, sometimes multiple
times, based on collected information and business logic. The
ability to pass along UUI in these call redirection scenarios is
critical.
2.4. Referral
In this scenario, the application uses a UA to initiate a referral,
which causes an INVITE request to be generated between the
originating UA and terminating UA with UUI data inserted by the
referrer UA. Note that this REFER method [RFC3515] could be part of
a transfer operation, or it might be unrelated to an existing call,
such as out-of-dialog REFER request. In some cases, this call flow
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is used in place of the redirection call flow: the referrer
immediately answers the call and then sends the REFER request. This
scenario is shown in Figure 3.
Originating UA Referrer Terminating UA
| | |
| REFER (UUI) F1 | |
|<-------------------| |
| 202 Accepted F2 | |
|------------------->| |
| INVITE (UUI) F3 | |
|---------------------------------------->|
| NOTIFY (100 Trying) F4 |
|------------------->| |
| 200 OK F5 | |
|<-------------------| |
| 200 OK F6 |
|<----------------------------------------|
| ACK F7 |
|---------------------------------------->|
| NOTIFY (200 OK) F8 | |
|------------------->| |
| 200 OK F9 | |
|<-------------------| |
Figure 3: Call Flow with Referral and UUI
3. Requirements
This section states the requirements for the transport of call
control User-to-User Information (UUI).
REQ-1: The mechanism will allow UAs to insert and receive UUI data in
SIP call setup requests and responses.
SIP messages covered by this include INVITE requests and end-to-
end responses to the INVITE, i.e., 18x and 200 responses. UUI
data may also be inserted in 3xx responses to an INVITE. However,
if a 3xx response is recursed on by an intermediary proxy, the
resulting INVITE will not contain the UUI data from the 3xx
response. In a scenario where a proxy forks an INVITE to multiple
UAS who include UUI data in 3xx responses, if a 3xx response is
the best response sent upstream by the proxy, it will contain the
UUI data from only one 3xx response.
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REQ-2: The mechanism will allow UAs to insert and receive UUI data in
SIP dialog terminating requests and responses.
Q.931 UUI supports inclusion in release and release completion
messages. SIP messages covered by this include BYE and 200 OK
responses to a BYE.
REQ-3: The mechanism will allow UUI to be inserted and retrieved in
SIP redirects and referrals.
SIP messages covered by this include REFER requests and 3xx
responses to INVITE requests.
REQ-4: The mechanism will allow UUI to be able to survive proxy
retargeting or redirection of the request.
Retargeting is a common method of call routing in SIP and must not
result in the loss of User-to-User Information.
REQ-5: The mechanism should not require processing entities to
dereference a URL in order to retrieve the UUI data.
Passing a pointer or link to the UUI data will not meet the real-
time processing considerations and would complicate interworking
with the PSTN.
REQ-6: The mechanism will support interworking with call-control-
related DSS1 information elements or QSIG information elements and
ISUP parameters.
REQ-7: The mechanism will allow a UAC to learn that a UAS understands
the UUI mechanism.
REQ-8: The mechanism will allow a UAC to require that a UAS
understands the call control UUI mechanism and have a request routed
based on this information. If the request cannot be routed to a UAS
that understands the UUI mechanism, the request will fail.
This could be useful in ensuring that a request destined for the
PSTN is routed to a gateway that supports the UUI mechanism rather
than an otherwise equivalent PSTN gateway that does not support
the ISDN mechanism. Note that support of the UUI mechanism does
not, by itself, imply that a particular application is supported
(see REQ-10).
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REQ-9: The mechanism will allow proxies to remove a particular
application usage of UUI data from a request or response.
This is a common security function provided by border elements to
header fields such as Alert-Info or Call-Info URIs. There is no
requirement for UAs to be able to determine if a particular usage
of UUI data has been removed from a request or response.
REQ-10: The mechanism will provide the ability for a UA to discover
which application usages of UUI another UA understands or supports.
The creation of a registry of application usages for the UUI
mechanism is implied by this requirement. The ISDN service
utilizes a field known as the protocol discriminator, which is the
first octet of the ISDN UUI data, for this purpose.
REQ-11: The UUI is a sequence of octets. The solution will provide a
mechanism of transporting at least 128 octets of user data and a one-
octet protocol discriminator, i.e., 129 octets in total.
There is the potential for non-ISDN services to allow UUI to be
larger than 128 octets. However, users of the mechanism will need
be cognizant of the size of SIP messages and the ability of
parsers to handle extremely large values.
REQ-12: The recipient of UUI will be able to determine the entity
that inserted the UUI. It is acceptable that this is performed
implicitly where it is known that there is only one other end UA
involved in the dialog. Where that does not exist, some other
mechanism will need to be provided. The UUI mechanism does not
introduce stronger authorization requirements for SIP; instead, the
mechanism needs to be able to utilize existing SIP approaches for
request and response identity.
This requirement comes into play during redirection, retargeting,
and referral scenarios.
4. Security Considerations
The security requirements for the UUI mechanism are described in this
section. It is important to note that UUI security is jointly
provided at the application layer and at the SIP layer. As such, is
important for application users of the UUI mechanism to know the
level of security used and deployed in their particular SIP
environments and not to assume that a standardized (but perhaps
rarely deployed) security mechanism is in place.
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There are three main security models that need to be addressed by the
UUI mechanism. One model treats the SIP layer as untrusted and
requires end-to-end integrity protection and/or encryption. This
model can be achieved by providing these security services at a layer
above SIP. In this case, the application integrity protects and/or
encrypts the UUI data before passing it to the SIP layer. This
method has two advantages: it does not assume or rely on end-to-end
security mechanisms in SIP, which have virtually no deployment, and
it allows an application that understands the contents of the UUI to
apply a proper level of security.
The second approach is for the application to pass the UUI without
any protection to the SIP layer and require the SIP layer to provide
this security. This approach is possible in theory, although its
practical use would be extremely limited.
The third model utilizes a trust domain and relies on perimeter
security at the SIP layer. This is the security model of the PSTN
and ISDN where UUI is commonly used today. This approach uses hop-
by-hop security mechanisms and relies on border elements for
filtering and application of policy. This approach is used today in
UUI deployments. Within this approach, there is a requirement that
intermediary elements can detect and remove a UUI element based on
policy, but there is no requirement that an intermediary element be
able to read or interpret the UUI (as the UUI contents only have end-
to-end significance).
The next three requirements capture the UUI security requirements.
REQ-13: The mechanism will allow integrity protection of the UUI.
This allows the UAS to be able to know that the UUI has not been
modified or tampered with by intermediaries. Note that there are
tradeoffs between this requirement and requirement REQ-9 for
proxies and border elements to remove UUI. One possible way to
satisfy both of these requirements is to utilize hop-by-hop
protection. This property is not guaranteed by the protocol in
the ISDN application.
REQ-14: The mechanism will allow end-to-end privacy of the UUI.
Some UUI may contain private or sensitive information and may
require different security handling from the rest of the SIP
message. Note that this property is not available in the ISDN
application.
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REQ-15: The mechanism will allow both end-to-end and hop-by-hop
security models.
The hop-by-hop model is required by the ISDN UUI service.
5. Acknowledgements
Thanks to Joanne McMillen, who was a co-author of earlier draft
versions of this specification. Thanks to Spencer Dawkins, Keith
Drage, Dale Worley, and Vijay Gurbani for their review of earlier
draft versions of this document. The authors wish to thank Christer
Holmberg, Frederique Forestie, Francois Audet, Denis Alexeitsev, Paul
Kyzivat, Cullen Jennings, and Mahalingam Mani for their comments on
this topic.
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.
[Q931] ITU-T, "ISDN user-network interface layer 3 specification
for basic call control", ITU-T Recommendation Q.931,
<http://www.itu.int/rec/T-REC-Q.931-199805-I/en>.
[Q763] ITU-T, "Signalling System No. 7 - ISDN User Part formats
and codes", ITU-T Recommendation Q.763,
<http://www.itu.int/rec/T-REC-Q.763-199912-I/en>.
[RFC6086] Holmberg, C., Burger, E., and H. Kaplan, "Session
Initiation Protocol (SIP) INFO Method and Package
Framework", RFC 6086, January 2011.
[Q933] ITU-T, "ISDN Digital Subscriber Signalling System No. 1
(DSS1) - Signalling specifications for frame mode switched
and permanent virtual connection control and status
monitoring", ITU-T Recommendation Q.933,
<http://www.itu.int/rec/T-REC-Q.933/en>.
[QSIG] ECMA, "Private Integrated Services Network (PISN) -
Circuit Mode Bearer Services - Inter-Exchange Signalling
Procedures and Protocol (QSIG-BC)", Standard ECMA-143,
December 2001.
[ANSI] ANSI, "Telecommunications-Integrated Services Digital
Network (ISDN)-Explicit Call Transfer Supplementary
Service", ANSI T1.643-1995.
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[ETSI] ETSI, "Integrated Services Digital Network (ISDN);
Diversion supplementary services", ETSI ETS 300 207-1, Ed.
1, 1994.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
Authors' Addresses
Alan Johnston
Avaya
St. Louis, MO 63124
EMail: alan.b.johnston@gmail.com
Laura Liess
Deutsche Telekom AG
EMail: laura.liess.dt@gmail.com
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