SIPPING WG A. Johnston, Ed.
Internet-Draft J. McMillen
Intended status: Informational Avaya
Expires: May 28, 2009 November 24, 2008
Transporting User to User Call Control Information in SIP for ISDN
Interworking
draft-johnston-sipping-cc-uui-06
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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 extension will also be used for native SIP endpoints
implementing similar services and interworking with ISDN services.
This document discusses requirements and approaches and recommends a
new header field be standardized. Example use cases include an
exchange between two user agents, retargeting by a proxy, and
redirection. An example application is in an Automatic Call
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Distributor (ACD) in a contact center.
Table of Contents
1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
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 . . . . . . . . . . . . . . . . . . . . . . . . . 7
5. Possible Mechanisms . . . . . . . . . . . . . . . . . . . . . 8
5.1. Why INFO is Not Used . . . . . . . . . . . . . . . . . . . 9
5.2. MIME body Approach . . . . . . . . . . . . . . . . . . . . 9
5.3. URI Parameter . . . . . . . . . . . . . . . . . . . . . . 9
5.4. Header Field Approach . . . . . . . . . . . . . . . . . . 10
6. Recommendation . . . . . . . . . . . . . . . . . . . . . . . . 11
7. Appendix - Syntax for UUI Header Field . . . . . . . . . . . . 11
7.1. IANA Considerations . . . . . . . . . . . . . . . . . . . 12
7.1.1. Registration of Header Field . . . . . . . . . . . . . 12
7.1.2. Registration of Header Field Parameter . . . . . . . . 13
7.1.3. Registration of SIP Option Tag . . . . . . . . . . . . 13
8. Security Considerations . . . . . . . . . . . . . . . . . . . 13
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
10. Informative References . . . . . . . . . . . . . . . . . . . . 14
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15
Intellectual Property and Copyright Statements . . . . . . . . . . 16
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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] data in SIP.
Q.931 UUI is widely used in the PSTN today in contact center 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
not able or 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.
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 discussed. Then, three mechanisms are
discussed and compared. The Appendix contains a header field
definition which meets all the requirements.
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
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[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 that 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 (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.
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 will allow the UUI to be transported directly in
the SIP message.
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Passing a pointer or link to the UUI information will not meet the
real-time processing considerations.
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.
4. Use Cases
This section discusses four uses 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 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.
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.
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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.
Originator Redirect Server Terminator
| | |
| INVITE F1 | |
|------------------->| |
| 302 Moved (UUI) F2 | |
|<-------------------| |
| ACK F3 | |
|------------------->| |
| INVITE (UUI) F4 | |
|---------------------------------------->|
| 200 OK F5 |
|<----------------------------------------|
| ACK F6 |
|---------------------------------------->|
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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.
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.
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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.
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. Possible Mechanisms
Three possible mechanisms for transporting UUI will be described:
MIME body, URI parameter, and header field transport.
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5.1. Why INFO is Not Used
Since the INFO method [RFC2976]., was developed for ISUP interworking
of user-to-user information, it might seem to be the logical choice
here. For non-call control user-to-user information, INFO can be
utilized for end to end transport. However, for transport of call
control user-to-user information, INFO can not be used. As the call
flows in the previous section show, the information is related to an
attempt to establish a session and must be passed with the session
setup request (INVITE), responses to that INVITE, or session
termination requests. As a result, it is not possible to use INFO in
these cases.
5.2. MIME body Approach
One method of transport is to transport the UUI information as a MIME
body. This is in keeping with the SIP-T architecture [RFC3372] in
which MIME bodies are used to transport ISUP information. Since the
INVITE will normally have an SDP message body, the resulting INVITE
with SDP and UUI will be multipart MIME. The insertion of a UUI
message body by a redirect server or in a REFER is difficult. The
body would need to be encoded in the Contact URI of the 3xx response
or the Refer-To URI of a REFER. For example:
Contact:
Note that the tag convention from SIP Torture Test
Messages [RFC4475] is used to show that there are no line breaks in
the actual message syntax.
The MIME body approach meets REQs 1-5. However, it does not meet
REQ-6 as support for Multipart MIME and escaped bodies in URIs is
uncommon in SIP UAs.
5.3. URI Parameter
Another proposed approach is to encode the UUI as a URI parameter
into the Contact or Refer-To URI.
Contact:
An INVITE sent to this Contact URI would contain UUI in the Request-
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URI of the INVITE. The URI parameter has a drawback in that a URI
parameter will not survive retargeting by a proxy as shown in Figure
2. That is, if the URI is included with an Address of Record instead
of a Contact URI, the URI parameter will not be copied over to the
Contact URI, resulting in the loss of the information. As a result,
this approach does not meet REQ-4.
5.4. Header Field Approach
Another approach that has been proposed is to use a header field to
transport the UUI information. The header field would be included in
INVITE requests and responses and BYE requests and responses, and
would pass transparently through proxies. For redirection, the
header field would be escaped into the Contact or Refer-To URI. This
is commonly supported in UAs due to call transfer use cases. As a
result, the header field approach supports REQs 1-6. In order to
meet REQ-7, a SIP feature tag is needed which can be included in
Supported and Require header fields.
The Call-Info header field is related to the UUI information.
However, there are a number of important differences:
Call-Info is typically used for rendering to the user. While some
of the UUI information may ultimately be rendered to the user,
most of the UUI information will be consumed by the end device.
Call-Info usually contains a URI pointer the information instead
of the actual information itself which does not REQ-5. It could
be possible to use a data URI to carry the UUI directly in this
header field.
The use of Call-Info for interworking to and from ISDN networks
seems problematic.
Overall, the overloading of the Call-Info header field for carrying
interworked UUI does not seem like a good idea. A separate header
field allows for clear policy and authorization rules to be used.
For these reasons, a separate header field needs to be defined,
described here as User-to-User. For example, here is an example
User-to-User header field from message F1 in Figure 1:
User-to-User: 56a390f3d2b7310023a;encoding=hex
For example, here is an escaped User-to-User header field from the
redirection response F2 of Figure 3:
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Contact:
The resulting INVITE F5 would contain the User-to-User header field
of the previous example.
An escaped User-to-User header field from the REFER message response
F1 of Figure 4:
Refer-To:
This would result in the INVITE F4 containing the User-to-User header
field of the first example.
6. Recommendation
The recommendation is to define a new SIP header field "User-to-User"
to transport UUI information in ISDN interworking applications since
this mechanism best supports the requirements. A SIP feature tag
"uui" also needs to be defined so that it can be used in Supported
and Require header fields to meet REQ-6.
The format of the UUI information is a topic of future
standardization. Currently, UUI is proprietary, requiring
coordinated configuration between servers. Standardizing the format
or providing content tags would provide additional benefits.
7. Appendix - Syntax for UUI Header Field
Editor's Note: Eventually this text will move to a SIP Working Group
document to define the new header field.
The User-to-User header field can be present in INVITE requests and
responses only and in BYE requests and responses.
Current usage is to interoperate with ISDN User to User Signaling
(UUS), a supplementary service in which manufacturer specific
information is transported via the codeset 0 User-user Information
IE. Three services are defined: service 1, service 2, and service 3.
This draft only addresses the SIP equivalent of service 1 although it
could easily be expanded later to address services 2 and 3. UUS
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Service 1 involves user to user signaling exchanged during call setup
and clearing within the following Q.931 call control messages: SETUP,
ALERT, CONNECT, DISCONNECT, RELEASE, and RELEASE COMPLETE. UUS
Service 2 involves user to user signaling exchanged during call
establishment (between ALERT and CONNECT) via the USER INFORMATION
message. This service usually has a maximum of 2 USER INFORMATION
messages in each direction. UUS Service 3 involves user to user
signaling exchanged on an active call via the USER INFORMATION
message.
The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC 2234 and extends RFC 3261.
UUI = "User-to-User" HCOLON uui-data *(SEMI uui-param)
uui-data = token
uui-param = enc-param | generic-param
enc-param = "encoding=" ("hex" | token)
Only one User-to-User header field may be present in a request or
response.
The only defined parameter for the User-to-User header field is the
encoding parameter. "encoding=hex" is used to indicate that the UUI
information is encoded as hex digits per the ISDN specification. The
first octet is the protocol discriminator). Other encoding methods
of encoding MAY also be standardized.
The UUI data MUST be less than 129 octets in length. This is because
ISDN limits UUI to 128 octets in length plus the single octet
protocol discriminator. Transporting UUI longer than 128 octets will
result in interoperability failures when interworking with ISDN.
7.1. IANA Considerations
7.1.1. Registration of Header Field
This document defines a new SIP header field named "User-to-User".
The following row shall be added to the "Header Fields" section of
the SIP parameter registry:
+------------------+--------------+-----------+
| Header Name | Compact Form | Reference |
+------------------+--------------+-----------+
| User-to-User | | [RFCXXXX] |
+------------------+--------------+-----------+
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Editor's Note: [RFCXXXX] should be replaced with the designation of
the eventual SIP Working Group document.
7.1.2. Registration of Header Field Parameter
This document defines a parameter for the header field defined in the
preceding section. The header field "User-to-User" can contain the
parameter "encoding".
The following rows shall be added to the "Header Field Parameters and
Parameter Values" section of the SIP parameter registry:
+------------------+----------------+-------------------+-----------+
| Header Field | Parameter Name | Predefined Values | Reference |
+------------------+----------------+-------------------+-----------+
| User-to-User | encoding | No | [RFCXXXX] |
+------------------+----------------+-------------------+-----------+
Editor's Note: [RFCXXXX] should be replaced with the designation of
the eventual SIP Working Group document.
7.1.3. Registration of SIP Option Tag
This specification registers a new SIP option tag, as per the
guidelines in Section 27.1 of [RFC3261].
This document defines the SIP option tag "uui".
The following row has been added to the "Option Tags" section of the
SIP Parameter Registry:
+------------+------------------------------------------+-----------+
| Name | Description | Reference |
+------------+------------------------------------------+-----------+
| uui | This option tag is used to indicate that | [RFCXXXX] |
| | a UA supports and understands the | |
| | User-to-User header field. | |
+------------+------------------------------------------+-----------+
Editor's Note: [RFCXXXX] should be replaced with the designation of
the eventual SIP Working Group document.
8. 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 contain private information
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that would require confidentiality and message integrity. Standard
SIP security mechanisms can be used to secure this header field. For
example, TLS transport can provide these properties over a single
hop. For multiple hop or end-to-end confidentiality and integrity
protection, S/MIME can be utilized.
Received User-to-User information should only be trusted if it is
authenticated or 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,
appropriate policy should be applied based on the PSTN trust domain.
9. Acknowledgements
The authors wish to thank Francois Audet, Denis Alexeitsev, Keith
Drage, Paul Kyzivat, and Mahalingam Mani for their comments.
10. 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 .
[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
Method", RFC 3515, April 2003.
[RFC3324] Watson, M., "Short Term Requirements for Network Asserted
Identity", RFC 3324, November 2002.
[RFC4475] Sparks, R., Hawrylyshen, A., Johnston, A., Rosenberg, J.,
and H. Schulzrinne, "Session Initiation Protocol (SIP)
Torture Test Messages", RFC 4475, May 2006.
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Authors' Addresses
Alan Johnston (editor)
Avaya
St. Louis, MO 63124
Email: alan@sipstation.com
Joanne McMillen
Avaya
Email: joanne@avaya.com
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this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
Johnston & McMillen Expires May 28, 2009 [Page 16]