CUSS WG A. Johnston
Internet-Draft Avaya
Intended status: Informational L. Liess
Expires: November 29, 2011 Deutsche Telekom AG
May 28, 2011

Problem Statement and Requirements for Transporting User to User Call Control Information in SIP


This document introduces the transport of call control related 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. A common example in another protocol is the ISDN User to User Information Service. 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 to achieve this. In addition, the 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.

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Table of Contents

1. Overview

This document describes the transport of User to User Information (UUI) during session setup. This section introduces UUI and explains how it relates to SIP.

We define SIP UUI information as application-specific information that is related to a session being established using SIP. It is assumed that the application is running in both the originator of the session and the terminator of the session. That is, the application interacts with the User Agent Client (UAC) and User Agent Server (UAS). 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).

UUI is defined this way for two reasons. Firstly, this 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. Secondly, it is neither feasible nor desirable 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 which 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 (UU IE) [Q931] and ITU-T Q.763 User to User Information Parameter [Q763] data in SIP. ISDN UUI is widely used in the 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 information between UAs during a session or dialog - the SIP INFO [RFC2976] 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 extensions.

In this document, four different use case call flows are discussed. Next, the requirements for call control UUI transport are discussed.

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. 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 will help motivate the requirements for SIP call control UUI.

3.1. 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 DSS1 information element or QSIG information element or 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.

           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.

3.2. 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.

The UUI in the INVITE needs to be passed unchanged through this proxy retargeting operation.

3.3. Redirection

In this scenario, UUI is inserted by an application which utilizes a SIP 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 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. However, this should not prevent the usage of SIP call control UUI in these cases. Instead, these slight differences between the SIP UUI service and the ISDN service need to be carefully noted and discussed in an interworking specification.

Figure 3 shows this scenario, with the Redirect inserting UUI which is then included in the INVITE F4 send to the Terminator.

           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.

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.

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 Terminator. The UUI could be used to lookup information rendered 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 maintain UUI in these scenarios is critical.

3.4. Referral

In this scenario, the application uses a UA to initiate a referral, which 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 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 Referral and UUI.

4. Requirements

This section discusses the requirements for the transport of call control related 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.

REQ-2: The mechanism will allow UAs to insert and receive UUI data in SIP dialog terminating requests and responses.

REQ-3: The mechanism will allow UUI to be inserted and retrieved in SIP redirects and referrals.

REQ-4: The mechanism will allow UUI to be able to survive proxy retargeting or any other form of redirection of the request.

REQ-5: The mechanism should not require processing entities to dereference a URL in order to retrieve the UUI information.

REQ-6 has been deleted. To avoid confusion, the number will not be reused.

REQ-7: The mechanism will support interworking with call control related DSS1 information elements or QSIG information elements or ISUP parameters.

REQ-8: The mechanism will allow a UAC to learn or request that a UAS understands the call control UUI mechanism and permit routing of requests based on this.

REQ-9: The mechanism will allow proxies to remove a particular type of UUI information from a request or response.

REQ-10: The mechanism will provide the ability for a UA to discover which types or application usages of UUI another UA understands or supports.

REQ-11: 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.

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.

REQ-13: The mechanism will allow integrity protection of the UUI.

REQ-14: The mechanism will allow end-to-end privacy of the UUI.

5. Security Considerations

The security requirements for the SIP UUI mechanism are described in REQ-13 and REQ-14, providing integrity protection and/or privacy. In addition, there is an identity requirement in REQ-12, which relates to the ability for the UAS to know who inserted the UUI.

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 SIP UUI to know the realistic level of security used and deployed in SIP, and not assume that some rarely deployed SIP level security mechanism is in place.

6. Acknowledgements

Thanks to Joanne McMillen who was a co-author of earlier versions of this specification. Thanks to Spencer Dawkins, Keith Drage, and Vijay Gurbani for their review of earlier 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.

7. 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)", , .
[Q763] ITU-T Q.763 Signaling System No. 7 - ISDN user part formats and codes", , .
[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.
[RFC2976] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.
[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures", BCP 63, RFC 3372, September 2002.
[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.

Authors' Addresses

Alan Johnston Avaya St. Louis, MO 63124 EMail:
Laura Liess Deutsche Telekom AG EMail: