One document matched: draft-johnston-dispatch-sip-cc-uui-01.txt
Differences from draft-johnston-dispatch-sip-cc-uui-00.txt
DISPATCH WG A. Johnston, Ed.
Internet-Draft J. McMillen
Intended status: Informational Avaya
Expires: September 6, 2010 March 5, 2010
Requirements for Transporting User to User Call Control Information in
SIP for ISDN Interworking
draft-johnston-dispatch-sip-cc-uui-01
Abstract
Several approaches to transporting the ITU-T Q.931 User to User
Information Element (UU IE) data in SIP have been proposed. As
networks move to SIP it is important that applications requiring this
data can continue to function in SIP networks as well as the ability
to interwork with this ISDN service for end-to-end transparency.
This document discusses requirements and approaches. This extension
will also be used for native SIP endpoints implementing similar
services and interworking with ISDN services. Example use cases
include an exchange between two user agents, retargeting by a proxy,
and redirection. An example application is an Automatic Call
Distributor (ACD) in a contact center.
Status of this Memo
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This Internet-Draft will expire on September 6, 2010.
Copyright Notice
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Copyright (c) 2010 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
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Table of Contents
1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4.1. User Agent to User Agent . . . . . . . . . . . . . . . . . 5
4.2. Proxy Retargeting . . . . . . . . . . . . . . . . . . . . 6
4.3. Redirection . . . . . . . . . . . . . . . . . . . . . . . 6
4.4. Referral . . . . . . . . . . . . . . . . . . . . . . . . . 8
5. Security Considerations . . . . . . . . . . . . . . . . . . . 9
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 10
7. Informative References . . . . . . . . . . . . . . . . . . . . 10
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11
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1. Overview
This document describes the transport of User to User Information
(UUI) in ISDN interworking scenarios using SIP [RFC3261].
Specifically, we discuss the transport of call control related ITU-T
Q.931 User to User Information Element (UU IE) [Q931] and ITU-T Q.763
User to User Information Parameter [Q763] data in SIP. UUI is widely
used in the PSTN today in contact centers and call centers which are
transitioning away from ISDN to SIP. This extension will also be
used for native SIP endpoints implementing similar services and
interworking with ISDN services.
Part of the definition of this ISDN service is that the UUI
information is not known and understood by the ISDN network that
transports it. This is for two reasons. Firstly, this supports a
strict layering of protocols and data. Providing information and
understanding of the data to the transport layer would not provide
any benefits and instead could create cross layer coupling and
increase the complexity of the system. Secondly, either the
originator or terminator of the service might be a simple PSTN
gateway designed for scalability and lowest cost. As a result, it is
neither feasible nor desirable for this device to understand the
information but instead the goal is to pass the information as
efficiently as possible to another application which does understand
the data. Both of these arguments still apply to SIP, especially
when one or both endpoints are gateways.
In the future, where both endpoints are intelligent SIP user agents,
it may be possible for them to understand and interpret the UUI data.
There may be some cases where the UUI information is relevant to SIP.
In this case, it might be worthwhile attempting to map UUI data to an
appropriate SIP header field or to standardize a new header field, or
to mark the UUI data as to the content and purpose. However, the
requirements and use cases for this are different enough from those
described in this document that these two situations should be
examined separately. This document looks only at the requirements
and mechanisms for replicating the existing, widely used and deployed
ISDN UUI service.
First, the requirements are discussed with use cases. Five different
use case call flows are then discussed.
The goal of this document is to generate discussions within the
DISPATCH Working Group for a charter for a new RAI area Working Group
to define a mechanism to meet these requirements and use cases.
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2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC 2119
[RFC2119].
3. Requirements
This section discusses the requirements for the transport of call
control related user to user information (UUI). We define call
control UUI as information that is generated, transported, and
consumed at the time of call setup (i.e. during a pending INVITE
transaction). The information can be used for call routing,
alerting, call distribution, or simply rendering. The exact usage
and semantics of call control UUI is out of scope - SIP is simply
providing the transport function for this, in the same manner as ISDN
Provides in the PSTN. Non-call control UUI can be sent using the
INFO method and not using the extensions described in this
specification.
REQ-1: The mechanism will allow user agents (UAs) to insert and
receive ITU-T Q.931 User to User Information Element and Q.763 User
to User Information Parameter (referred to as 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, which includes 18x and 200 responses.
REQ-2: The mechanism will allow UAs to insert and receive ITU-T Q.931
User to User Information Element (referred to as UUI) data in SIP
call teardown 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 to INVITEs.
SIP messages covered by this include 3xx responses to INVITE and
REFER requests.
REQ-4: The mechanism will allow UUI to be able to survive proxy
retargeting.
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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 to retrieve the UUI information.
Passing a pointer or link to the UUI information will not meet the
real-time processing considerations and will complicate
interworking with the PSTN.
REQ-6: The mechanism will minimize reliance on SIP extensions or
uncommon SIP behavior.
REQ-7: The mechanism will allow the inserter of UUI to be sure that
the recipient understands the call control UUI mechanism.
Understanding the mechanism means that the UAS will extract and
utilize the UUI information transported. Understanding the
protocol, format, and nature of the actual UUI data is not covered
by this requirement. Note that this requirement is not strictly
needed to implement the UUS 1 implicit service, but maps more
accurately to the UUS 1 explicit service. However, having an
option tag is good design for high reliability systems, and the
dynamic and heterogeneous nature of SIP interconnection (as
opposed to the PSTN's static trunking) makes this option tag much
more important and hence relevant to even the UUS 1 implicit
service.
4. Use Cases
This section discusses four use cases for the transport of call
control related user to user information. What is not discussed here
is the transport of non-call control UUI which can be done using the
SIP INFO method. These use cases help explain the requirements from
the previous section.
4.1. User Agent to User Agent
In this scenario, the originator UA includes UUI in the INVITE sent
through a proxy to the terminating UA. The terminator can use the
UUI in any way. If it is an ISDN gateway, it could map the UUI into
the appropriate Q.931 or Q.763 element. Alternatively, it might
render the information to the user, or use it for 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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Originator Proxy Terminator
| | |
| 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 Originator and
Terminator.
This call flow utilizes REQ-1.
4.2. Proxy Retargeting
In this scenario, the originator UA includes UUI in the INVITE sent
through a proxy to the terminating UA. The proxy retargets the
INVITE, sending it to a different termination UA. The UUI
information is then received and processed by the terminating UA.
This call flow is shown in Figure 2.
Originator Proxy Terminator 2
| | |
| INVITE (UUI) F1 | |
|------------------->| INVITE (UUI) F2 |
| 100 Trying F3 |------------------->|
|<-------------------| 200 OK F4 |
| 200 OK F5 |<-------------------|
|<-------------------| |
| ACK F6 | |
|------------------->| ACK F7 |
| |------------------->|
Figure 2. Call flow with Proxy Retargeting.
This call flow utilizes REQ-1 and REQ-4.
4.3. Redirection
In this scenario, UUI is inserted by a redirect server. The UUI is
then included in the INVITE sent by the Originator to the Terminator.
In this case, the Originator does not necessarily need to support the
UUI mechanism but does need to support the SIP redirection mechanism
used to include the UUI information. Two examples of UUI with
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redirection (transfer and diversion) are defined in [ANSII] 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.
In redirection scenarios, if the Redirect Server is not in the same
administrative domain as the Terminator, the Redirect Server must not
remove or replace any UUI in the initial INVITE. In Figure 3, this
means that if F1 included UUI, the Redirect Server could not modify
or replace the UUI in F2. However, if the Redirect Server and the
Terminator are part of the same administrative domain, they may have
a policy allowing the Redirect Server to modify or rewrite UUI
information. In fact, many UUI uses within an Enterprise rely on
this feature to work today in ISDN.
Originator Redirect Server Terminator
| | |
| INVITE F1 | |
|------------------->| |
| 302 Moved (UUI) F2 | |
|<-------------------| |
| ACK F3 | |
|------------------->| |
| INVITE (UUI) F4 | |
|---------------------------------------->|
| 200 OK F5 |
|<----------------------------------------|
| ACK F6 |
|---------------------------------------->|
Figure 3. Call flow with UUI exchanged between Redirect Server and
Terminator
This call flow utilizes REQ-1 and REQ-3.
A common 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 Terminator. The UUI could be used to
lookup information rendered to the agent at the time of call
answering.
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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 maintain UUI in these scenarios is critical.
4.4. Referral
In this scenario, a Referrer UA causes an INVITE to be generated
between the Originator and Terminator with UUI information inserted
by the Referrer UA. Note that this REFER [RFC3515] could be part of
a transfer operation or it might be unrelated to an existing call,
such as out-of-dialog REFER call control. In some cases, this call
flow is used in place of the redirection call flow, but where
immediately upon answer, the REFER is sent. This scenario is shown
in Figure 4.
Originator Referrer Terminator
| | |
| REFER (UUI) F1 | |
|<-------------------| |
| 202 Accepted F2 | |
|------------------->| |
| NOTIFY (100 Trying) F3 |
|------------------->| |
| 200 OK F4 | |
|<-------------------| |
| INVITE (UUI) F5 | |
|---------------------------------------->|
| 200 OK F6 |
|<----------------------------------------|
| ACK F7 |
|---------------------------------------->|
| NOTIFY (200 OK) F8 | |
|------------------->| |
| 200 OK F9 | |
|<-------------------| |
Figure 4. Call flow with transfer after answer.
Some scenarios involving referral have been proposed to use a REFER
sent during an early dialog. This NOT RECOMMENDED call flow is shown
in Figure 5. This flow is not recommended due to the number of
messages exchanged (due to the REFER, CANCEL, and 487 responses) and
the sending of the REFER in the early dialog. Also, there are race
conditions that can occur if a 200 OK to the INVITE is received by
the Originator while the REFER is in progress.
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Originator Referrer Terminator
| | |
| INVITE F1 | |
|------------------->| |
| 180 Ringing F2 | |
|<-------------------| |
| REFER (UUI) F3 | |
|<-------------------| |
| 202 Accepted F4 | |
|------------------->| |
| NOTIFY (100 Trying) F5 |
|------------------->| |
| 200 OK F6 | |
|<-------------------| |
| INVITE (UUI) F7 | |
|---------------------------------------->|
| 200 OK F8 |
|<----------------------------------------|
| ACK F9 |
|---------------------------------------->|
| NOTIFY (200 OK) F10| |
|------------------->| |
| 200 OK F11 | |
|<-------------------| |
| CANCEL F12 | |
|------------------->| |
| 200 OK F13 | |
|<-------------------| |
| 487 Request Terminated F14 |
|<-------------------| |
| ACK F15 | |
|------------------->| |
Figure 5. NOT RECOMMENDED call flow showing REFER prior to answer.
5. Security Considerations
User to user information can be exchanged over SIP on a hop-by-hop or
end-to-end basis. In some cases, UUI may carry privacy information
that would require confidentiality and message integrity. Standard
SIP security mechanisms, viz., based on TLS, offer these properties
per-hop. To preserve multi-hop or end-end confidentiality and
integrity, S/MIME profile MUST be utilized. Since the security
requirements and key management of the UUI information are likely to
be quite different from the SIP signaling transport, another approach
would be for the UUI information to be encrypted before being passed
to SIP for transport.
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Received User-to-User information should only be trusted if it is
authenticated or if it is received within a trust domain. For
example, Spec-T, defined in [RFC3324] could be used to define a trust
domain. When utilized by a gateway to map information to or from
ISDN Q.931 and ISUP Q.763, appropriate policy should be applied based
on the PSTN trust domain.
6. Acknowledgements
Thanks to Spencer Dawkins, Keith Drage, Vijay Gurbani, and Laura
Liess for their review of the document. The authors wish to thank
Francois Audet, Denis Alexeitsev, Paul Kyzivat, Cullen Jennings, and
Mahalingam Mani for their comments.
7. 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 Q.931 User to User Information Element (UU IE)",
http://www.itu.int/rec/T-REC-Q.931-199805-I/en .
[Q763] "ITU-T Q.763 Signaling System No. 7 - ISDN user part
formats and codes",
http://www.itu.int/rec/T-REC-Q.931-199805-I/en .
[ANSII] "ANSI T1.643-1995, Telecommunications-Integrated Services
Digital Network (ISDN)-Explicit Call Transfer
Supplementary Service".
[ETSI] "ETSI ETS 300 207-1 Ed.1 (1994), Integrated Services
Digital Network (ISDN); Diversion supplementary services".
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation Protocol
for Telephones (SIP-T): Context and Architectures",
BCP 63, RFC 3372, September 2002.
[RFC2976] Donovan, S., "The SIP INFO Method", RFC 2976,
October 2000.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer
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Method", RFC 3515, April 2003.
[RFC3324] Watson, M., "Short Term Requirements for Network Asserted
Identity", RFC 3324, November 2002.
Authors' Addresses
Alan Johnston (editor)
Avaya
St. Louis, MO 63124
Email: alan.b.johnston@gmail.com
Joanne McMillen
Avaya
Email: joanne@avaya.com
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