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rfc:rfc3976

Network Working Group V. K. Gurbani Request for Comments: 3976 Lucent Technologies, Inc. Category: Informational F. Haerens

                                                          Alcatel Bell
                                                            V. Rastogi
                                                    Wipro Technologies
                                                          January 2005
     Interworking SIP and Intelligent Network (IN) Applications

Status of This Memo

 This memo provides information for the Internet community.  It does
 not specify an Internet standard of any kind.  Distribution of this
 memo is unlimited.

Copyright Notice

 Copyright (C) The Internet Society (2005).

IESG Note

 This RFC is not a candidate for any level of Internet Standard.  The
 IETF disclaims any knowledge of the fitness of this RFC for any
 purpose, and in particular notes that the decision to publish is not
 based on IETF review for such things as security, congestion control,
 or inappropriate interaction with deployed protocols.  The RFC Editor
 has chosen to publish this document at its discretion.  Readers of
 this document should exercise caution in evaluating its value for
 implementation and deployment.  See RFC 3932 for more information.

Abstract

 Public Switched Telephone Network (PSTN) services such as 800-number
 routing (freephone), time-and-day routing, credit-card calling, and
 virtual private network (mapping a private network number into a
 public number) are realized by the Intelligent Network (IN).  This
 document addresses means to support existing IN services from Session
 Initiation Protocol (SIP) endpoints for an IP-host-to-phone call.
 The call request is originated on a SIP endpoint, but the services to
 the call are provided by the data and procedures resident in the
 PSTN/IN.  To provide IN services in a transparent manner to SIP
 endpoints, this document describes the mechanism for interworking SIP
 and Intelligent Network Application Part (INAP).

Gurbani, et al. Informational [Page 1] RFC 3976 Interworking SIP & IN January 2005

Table of Contents

 1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
 2.  Access to IN-Services from a SIP Entity. . . . . . . . . . . .  4
 3.  Additional SIN Considerations  . . . . . . . . . . . . . . . .  7
     3.1.  The Concept of State in SIP. . . . . . . . . . . . . . .  7
     3.2.  Relationship between SCP and a SIN-Enabled SIP entity. .  7
     3.3.  SIP REGISTER and IN services . . . . . . . . . . . . . .  8
     3.4.  Support of Announcements and Mid-Call Signaling. . . . .  8
 4.  The SIN Architecture . . . . . . . . . . . . . . . . . . . . .  8
     4.1.  Definitions. . . . . . . . . . . . . . . . . . . . . . .  8
     4.2.  IN Service Control Based on the SIN Approach . . . . . .  9
 5.  Mapping of the SIP State Machine to the IN State Model . . . . 10
     5.1.  Mapping SIP Protocol State Machine to O_BCSM . . . . . . 11
     5.2.  Mapping SIP Protocol State Machine to T_BCSM . . . . . . 16
 6.  Example Call Flows . . . . . . . . . . . . . . . . . . . . . . 20
 7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 21
 8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
     8.1.  Normative References . . . . . . . . . . . . . . . . . . 21
     8.2.  Informative References . . . . . . . . . . . . . . . . . 22
     Appendix A . . . . . . . . . . . . . . . . . . . . . . . . . . 23
     Acknowledgments. . . . . . . . . . . . . . . . . . . . . . . . 24
     Author's Addresses . . . . . . . . . . . . . . . . . . . . . . 24
     Full Copyright Statement . . . . . . . . . . . . . . . . . . . 25

1. Introduction

 PSTN services such as 800-number routing (freephone), time-and-day
 routing, credit-card calling, and virtual private network (mapping a
 private network number into a public number) are realized by the
 Intelligent Network.  IN is an architectural concept for the real-
 time execution of network services and customer applications [1].  IN
 is, by design, de-coupled from the call processing component of the
 PSTN.  In this document, we describe the means to leverage this
 decoupling to provide IN services from SIP-based entities.
 First, we will explain the basics of IN.  Figure 1 shows a simplified
 IN architecture, in which telephone switches called Service Switching
 Points (SSPs) are connected via a packet network called Signaling
 System No. 7 (SS7) to Service Control Points (SCPs), which are
 general purpose computers.  At certain points in a call, a switch can
 interrupt a call and request instructions from an SCP on how to
 proceed with the call.  The points at which a call can be interrupted
 are standardized within the Basic Call State Model (BCSM) [1, 2].
 The BCSM models contain two processes, one each for the originating
 and terminating part of a call.

Gurbani, et al. Informational [Page 2] RFC 3976 Interworking SIP & IN January 2005

 When the SCP receives a request for instructions, it can reply with a
 single response, such as a simple number translation augmented by
 criteria like time of day or day of week, or, in turn, initiate a
 complex dialog with the switch.  The situation is further complicated
 by the necessity to engage other specialized devices that collect
 digits, play recorded announcements, perform text-to-speech or
 speech-to-text conversions, etc.  (These devices are not discussed
 here.)  The related protocol, as well as the BCSM, is standardized by
 the ITU-T and known as the Intelligent Network Application Part
 protocol (INAP) [4].  Only the protocol, not an SCP API, has been
 standardized.
                        +-----------+
                        |           |
                        |    SCP    |
                        |           |
                        +-----------+
                              ||
                              ||
                             /  \
                            /    \
                           / INAP \
                          /        \
                         /          \
                +--------+  ISUP   +--------+
                |  SSP   |*********|  SSP   |
                +--------+         +--------+
                Figure 1.  Simplified IN Architecture
 The overall objective is to ensure that IN control of Voice over IP
 (VoIP) services in networks can be readily specified and implemented
 by adapting standards and software used in the present networks.
 This approach leads to services that function the same when a user
 connects to present or future networks, simplifies service evolution
 from present to future, and leads to more rapid implementation.
 The rest of this document is organized as follows: Section 2 contains
 the architectural model of an IN aware SIP entity.  Section 3
 provides some issues to be taken into account when performing SIP/IN
 interworking (SIN).  Section 4 discusses the IN service control based
 on the SIN approach.  The technique outlined in this document focuses
 on the call models of IN and the SIP protocol state machine; Section
 5 thus establishes a complete mapping between the two state machines
 that allows access to IN services from SIP endpoints.  Section 6
 includes call flows of IN services executing on SIP endpoints.  These
 services are readily enabled by the technique described in this
 document.  Finally, Section 7 covers security aspects of SIN.

Gurbani, et al. Informational [Page 3] RFC 3976 Interworking SIP & IN January 2005

List of Acronyms

 B2BUA       Back-to-Back User Agent
 BCSM        Basic Call State Model
 CCF         Call Control Function
 DP          Detection Point
 DTMF        Dual Tone Multi-Frequency
 IN          Intelligent Network
 INAP        Intelligent Network Application Part
 IP          Internet Protocol
 ITU-T       International Telecommunications Union -
             Telecommunications Standardization Sector
 O_BCSM      Originating Basic Call State Model
 PIC         Point in Call
 PSTN        Public Switched Telephone Network
 RTP         Real Time Protocol
 R-URI       Request URI
 SCF         Service Control Function
 SCP         Service Control Point
 SIGTRAN     Signal Transport Working Group in IETF
 SIN         SIP/IN Interworking
 SIP         Session Initiation Protocol
 SS7         Signaling System  No. 7
 SSF         Service Switching Function
 SSP         Service Switching Point
 T_BCSM      Terminating Basic Call State Model
 UA          User Agent
 UAC         User Agent Client
 UAS         User Agent Server
 VoIP        Voice over IP
 VPN         Virtual Private Network

2. Access to IN-Services from a SIP Entity

 The intent of this document is to provide the means to support
 existing IN-based applications in a SIP [3] environment.  One way to
 gain access to IN services transparently from SIP (e.g., through the
 same detection points (DPs) and point-in-call (PIC) used by
 traditional switches) is to map the SIP protocol state machine to the
 IN call models [1].
 From the viewpoint of IN elements such as the SCP, the request's
 origin from a SIP entity rather than a call processing function on a
 traditional switch is immaterial.  Thus, it is important that the SIP
 entity be able to provide the same features as the traditional
 switch, including operating as an SSP for IN features.  The SIP
 entity should also maintain call state and trigger queries to IN-
 based services, as do traditional switches.

Gurbani, et al. Informational [Page 4] RFC 3976 Interworking SIP & IN January 2005

 This document does not intend to specify which SIP entity shall
 operate as an SSP; however, for the sake of completeness, it should
 be mentioned that this task should be performed by SIP entities at
 (or near) the core of the network rather than at the SIP end points
 themselves.  To that extent, SIP entities such as proxy servers and
 Back-to-Back user agents (B2BUAs) may be employed.  Generally
 speaking, proxy servers can be used for IN services that occur during
 a call setup and teardown.  For IN services requiring specialized
 media handling (such as DTMF detection) or specialized call control
 (such as placing parties on hold) B2BUAs will be required.
 The most expeditious manner for providing existing IN services in the
 IP domain is to use the deployed IN infrastructure as often as
 possible.  In SIP, the logical point to tap into for accessing
 existing IN services is either the user agents or one of the proxies
 physically closest to the user agent (and presumably in the same
 administrative domain).  However, SIP entities do not run an IN call
 model; to access IN services transparently, the trick then is to
 overlay the state machine of the SIP entity with an IN layer so that
 call acceptance and routing is performed by the native state machine
 and so that services are accessed through the IN layer by using an IN
 call model.  Such an IN-enabled SIP entity, operating in synchrony
 with the events occurring at the SIP transaction level and
 interacting with the IN elements (SCP), is depicted in Figure 2:
                      +-------+
                      | SCP   |
                      +---+---+
                          |
                          | INAP
                          |
                      +--------+
                      | SIN    |
                      +........+
                      |  SIP   |
           ---------->| Entity |--------->
           Requests   |        | Requests out
           in         +--------+ (after applying IN
                                  services)
          SIN: SIP/IN Interworking layer
          Figure 2.  SIP Entity Accessing IN Services
 Section 5 proposes this mapping between the IN layer and the SIP
 protocol state machine.  Essentially, a SIP entity exhibiting this
 mapping becomes a SIN-enabled SIP entity.

Gurbani, et al. Informational [Page 5] RFC 3976 Interworking SIP & IN January 2005

 This document does not propose any extensions to SIP.
 Figure 3 expands the SIP entity depicted in Figure 2 and further
 details the architecture model involving IN and SIP interworking.
 Events occurring at the SIP layer will be passed to the IN layer for
 service application.  More specifically, since IN services deal with
 E.164 numbers, it is reasonable to assume that a SIN-enabled SIP
 entity that seeks to provide services on such a number will consult
 the IN layer for further processing, thus acting as a SIP-based SSP.
 The IN layer will proceed through its BCSM states and, at appropriate
 points in the call, will send queries to the SCP for call
 disposition.  Once the disposition of the call has been determined,
 the SIP layer is informed and processes the transaction accordingly.
 Note that the single SIP entity as modeled in this figure can in fact
 represent several different physical instances in the network as, for
 example, when one SIP entity is in charge of the terminal or access
 network/domain, and another is in charge of the interface to the
 Switched Circuit Network (SCN).
                +-------+
                |  SCP  |
                +---o---+
                    |
                    +-----+
                          |
                **********|***********************************
                * +-------|-------------------+              *
                * |+------o------+            |              *
                * ||  SSF(IP)    |            |              *
                * |+-------------+            |              *
                * ||  CCF(IP)    |            |              *
                * |+------o------+            |              *
                * +-------|-------------------+              *
                *         |                      SIN-enabled *
                * +-------o-------------------+  SIP         *
                * |      SIP Layer            |  Entity      *
                * +---------------------------+              *
                **********************************************
   Figure 3.  Functional Architecture of a SIN-Enabled SIP Entity
 The following architecture entities, used in Figure 3, are defined in
 the Intelligent Network standards:
       Service Switching Function (SSF): IN functional entity that
       interacts with call control functions.

Gurbani, et al. Informational [Page 6] RFC 3976 Interworking SIP & IN January 2005

       Call Control Function (CCF): IN functional entity that refers
       to call and connection handling in the classical sense (i.e.,
       that of an exchange).

3. Additional SIN Considerations

 In working between Internet Telephony and IN-PSTN networks, the main
 issue is to translate between the states produced by the Internet
 Telephony signaling and those used in traditional IN environments.
 Such a translation entails attention to the considerations listed
 below.

3.1. The Concept of State in SIP

 IN services occur within the context of a call, i.e., during call
 setup, call teardown, or in the middle of a call.  SIP entities such
 as proxies, with which some of these services may be realized,
 typically run in transaction-stateful (or stateless) mode.  In this
 mode, a SIP proxy that proxied the initial INVITE is not guaranteed
 to receive a subsequent request, such as a BYE.  Fortunately, SIP has
 primitives to force proxies to run in a call-stateful mode; namely,
 the Record-Route header.  This header forces the user agent client
 (UAC) and user agent server (UAS) to create a "route set" that
 consists of all intervening proxies through which subsequent requests
 must traverse.  Thus SIP proxies must run in call-stateful mode in
 order to provide IN services on behalf of the UAs.
 A B2BUA is another SIP element in which IN services can be realized.
 As a B2BUA is a true SIP UA, it maintains complete call state and is
 thus capable of providing IN services.

3.2. Relationship between SCP and a SIN-Enabled SIP Entity

 In the architecture model proposed in this document, each SIN-enabled
 SIP entity is pre-configured to communicate with one logical SCP
 server, using whatever communication mechanism is appropriate.
 Different SIP servers (e.g., those in different administrative
 domains) may communicate with different SCP servers, so that there is
 no single SCP server responsible for all SIP servers.
 As Figures 1 and 2 depict, the IN-portion of the SIN-enabled SIP
 entity will communicate with the SCP.  This interface between the IN
 call handling layer and the SCP is not specified by this document
 and, indeed, can be any one of the following, depending on the
 interfaces supported by the SCP: INAP over IP, INAP over SIGTRAN, or
 INAP over SS7.

Gurbani, et al. Informational [Page 7] RFC 3976 Interworking SIP & IN January 2005

 This document is only applicable when SIP-controlled Internet
 telephony devices seek to operate with PSTN devices.  The SIP UAs
 using this interface would typically appear together with a media
 gateway.  This document is *not* applicable in an all-IP network and
 is not needed in cases where PSTN media gateways (not speaking SIP)
 need to communicate with SCPs.

3.3. SIP REGISTER and IN Services

 SIP REGISTER provisions a SIP Proxy or SIP Registration server.  The
 process is similar to the provisioning of an SCP/HLR in the switched
 circuit network.  SCPs that provide VoIP based services can leverage
 this information directly.  However, this document neither endorses
 nor prohibits such an architecture and, in fact, considers it an
 implementation decision.

3.4. Support of Announcements and Mid-Call Signaling

 Services in the IN such as credit-card calling typically play
 announcements and collect digits from the caller before a call is set
 up.  Playing announcements and collecting digits require the
 manipulation of media streams.  In SIP, proxies do not have access to
 the media data path.  Thus, such services should be executed in a
 B2BUA.
 Although the SIP specification [3] allows for end points to be put on
 hold during a call or for a change of media streams to take place, it
 does not have any primitives to transport other than mid-call control
 information.  This may include transporting DTMF digits, for example.
 Extensions to SIP, such as the INFO method [5] or the SIP event
 notification extension [6], can be considered for services requiring
 mid-call signaling.  Alternatively, DTMF can be transported in RTP
 itself [7].

4. The SIN Architecture

4.1. Definitions

 The SIP architecture has the following functional elements defined in
 [3]:
  1. User agent client (UAC): The SIP functional entity that

initiates a request.

  1. User agent server (UAS): The SIP functional entity that

terminates a request by sending 0 or more provisional SIP

       responses and one final SIP response.

Gurbani, et al. Informational [Page 8] RFC 3976 Interworking SIP & IN January 2005

  1. Proxy server: An intermediary SIP entity that can act as both a

UAS and a UAC. Acting as a UAS, it accepts requests from UACs,

       rewrites the Request-URI (R-URI), and, acting as a UAC, proxies
       the request to a downstream UAS.  Proxies may retain
       significant call control state by inserting themselves in
       future SIP transactions beyond the initial INVITE.
  1. Redirect server: An intermediary SIP entity that redirects

callers to alternate locations, after possibly consulting a

       location server to determine the exact location of the callee
       (as specified in the R-URI).
  1. Registrar: A SIP entity that accepts SIP REGISTER requests and

maintains a binding from a high-level URL to the exact location

       for a user.  This information is saved in some data-store that
       is also accessible to a SIP Proxy and a SIP Redirect server.  A
       Registrar is usually co-located with a SIP Proxy or a SIP
       Redirect server.
  1. Outbound proxy: A SIP proxy located near the originator of

requests. It receives all outgoing requests from a particular

       UAC, including those requests whose R-URIs identify a host
       other than the outbound proxy.  The outbound proxy sends these
       requests, after any local processing, to the address indicated
       in the R-URI.
  1. Back-to-Back UA (B2BUA): A SIP entity that receives a request

and processes it as a UAS. It also acts as a UAC and generates

       requests to determine how the incoming request is to be
       answered.  A B2BUA maintains complete dialog state and must
       participate in all requests sent within the dialog.

4.2. IN Service Control Based on the SIN Approach

 Figure 4 depicts the possibility of IN service control based on the
 SIN approach.  On both the originating and terminating ends, a SIN-
 capable SIP entity is assumed (it can be a proxy or a B2BUA).  The "O
 SIP" entity is required for outgoing calls that require support for
 existing IN services.  Likewise, on the callee's side (or terminating
 side), an equally configured entity ("T SIP") will be required to
 provide terminating side services.  Note that the "O SIP" and "T SIP"
 entities correspond, respectively, to the IN O_BCSM and T_BCSM halves
 of the IN call model.

Gurbani, et al. Informational [Page 9] RFC 3976 Interworking SIP & IN January 2005

   +---+                                                       +---+
   | S |                    (~~~~~~~~~~~~~)                    | S |
   | C |<--+               (               )               +-->| C |
   | P |   |              (                 )              |   | P |
   +---+   |             (   Switched        )             |   +---+
           |             (   Circuit         )             |
           V             (   Network         )             V
    +-------+            (                   )          +-------+
    | SIN   |    +---------+           +---------+      | SIN   |
    +-------+----| Gateway |    ...    | Gateway |------+-------+
    | O SIP |    +---------+           +---------+      | T SIP |
    +-------+             (                 )           +-------+
                           (               )
                            (.............)
   O SIP: Originating SIP entity
   T SIP: Terminating SIP entity
   Figure 4.  Overall SIN Architecture

5. Mapping of the SIP State Machine to the IN State Model

 This section establishes the mapping of the SIP protocol state
 machine to the IN generic basic call state model (BCSM) [2],
 independent of any capability sets [8, 9].  The BCSM is divided into
 two halves: an originating call model (O_BCSM) and a terminating call
 model (T_BCSM).  There are a total of 19 PICs and 35 DPs between both
 the halves (11 PICs and 21 DPs for O_BCSM; 8 PICs and 14 DPs for
 T_BCSM) [1].  The SSPs, SCPs, and other IN elements track a call's
 progress in terms of the basic call model.  The basic call model
 provides a common context for communication about a call.
 O_BCSM has 11 PICs:
 O_NULL: Starting state; call does not exist yet.
 AUTH_ORIG_ATTEMPT: Switch detects a call setup request.
 COLLECT_INFO: Switch collects the dial string from the calling party.
 ANALYZE_INFO: Complete dial string is translated into a routing
    address.
 SELECT_ROUTE: Physical route is selected, based on the routing
    address.
 AUTH_CALL_SETUP: Switch ensures the calling party is authorized to
    place the call.
 CALL_SENT: Control of call sent to terminating side.
 O_ALERTING: Switch waits for the called party to answer.
 O_ACTIVE: Connection established; communications ensue.
 O_DISCONNECT: Connection torn down.
 O_EXCEPTION: Switch detects an exceptional condition.

Gurbani, et al. Informational [Page 10] RFC 3976 Interworking SIP & IN January 2005

 T_BCSM has 8 PICS:
 T_NULL: Starting state; call does not exist yet.
 AUTH_TERM_ATT: Switch verifies whether the call can be sent to
    terminating party.
 SELECT_FACILITY: Switch picks a terminating resource to send the call
    on.
 PRESENT_CALL: Call is being presented to the called party.
 T_ALERTING: Switch alerts the called party, e.g., by ringing the
    line.
 T_ACTIVE: Connection established; communications ensue.
 T_DISCONNECT: Connection torn down.
 T_EXCEPTION: Switch detects an exceptional condition.
 The state machine for O_BCSM and T_BCSM is provided in [1] on pages
 98 and 103, respectively.  This state machine will be used for
 subsequent discussion when the IN call states are mapped into SIP.
 The next two sections contain the mapping of the SIP protocol state
 machine to the IN BCSMs.  Explaining all PICs and DPs in an IN call
 model is beyond the scope of this document.  It is assumed that the
 reader has some familiarity with the PICs and DPs of the IN call
 model.  More information can be found in [1].  For a quick reference,
 Appendix A contains a mapping of the DPs to the SIP response codes as
 discussed in the next two sections.

5.1. Mapping SIP Protocol State Machine to O_BCSM

 The 11 PICs of O_BCSM come into play when a call request (SIP INVITE
 message) arrives from an upstream SIP client to an originating SIN-
 enabled SIP entity running the IN call model.  This entity will
 create an O_BCSM object and initialize it in the O_NULL PIC.  The
 next seven IN PICs -- O_NULL, AUTH_ORIG_ATT, COLLECT_INFO,
 ANALYZE_INFO, SELECT_ROUTE, AUTH_CALL_SETUP, and CALL_SENT -- can all
 be mapped to the SIP "Calling" state.
 Figure 5 provides a visual map from the SIP protocol state machine to
 the originating half of the IN call model.  Note that control of the
 call shuttles between the SIP protocol machine and the IN O_BCSM call
 model while it is being serviced.

Gurbani, et al. Informational [Page 11] RFC 3976 Interworking SIP & IN January 2005

          SIP                                      O_BCSM
         | INVITE
         V
    +---------+                        +---------------+
    | Calling +=======================>+ O_NULL        +<----+
    +--+---/\-+                        +-/\---+--------+     |
    |  |   ||    +-------------+         |    |              |
    |  |   ||<===+O_Exception  +---------+ +--V-+         +--+-+
    |  |   ||    +--/\---------+           |DP 1|         |DP21|
    |  |   ||       |    +----+      +-----+----+------+  +--+-+
    |  |   ||       +<---+DP 2|<-----+ Auth_Orig._Att  +---->+
    |  |   ||       |    +----+      +--------+--------+     |
    |  |   ||       |                         |              |
    |  |   ||       |                      +--V-+            |
    |  |   ||       |                      |DP 3|            |
    |  |   ||       |    +----+      +-----+----+------+     |
    |  |   ||       +<---+DP 4|<-----+ Collect_Info    +---->+
    |  |   ||       |    +----+      +--------+--------+     |
    |  |   ||       |                         |              |
    |  |   ||       |                      +--V-+            |
    |  |   ||       |                      |DP 5|            |
    |  |   ||       |    +----+      +-----+----+------+     |
    |  |   ||       +<---+DP 6|<-----+ Analyze_Info    +---->+
    |  |   ||       |    +----+      +--------+--------+     |
    |  |   ||       |                         |              |
    |  |   ||       |                      +--V-+            |
    |  |   ||       |                      |DP 7|            |
    |  |   ||       |    +----+      +-----+----+------+     |
    |  |   ||       +<---+DP 8|<-----+ Select_Route    +---->+
    |  |   ||       |    +----+      +--------+--------+     |
    |  |   ||       |                         |              |
    |  |   ||       |                      +--V-+            |
    |  |   ||       |                      |DP 9|            |
    |  |   ||       |    +----+      +-----+----+------+     |
    |  |   ||       +<---+DP10|<-----+ Auth._Call_Setup+---->+
    |  |   ||            +----+      +--------+--------+

+—-+ | || | | | || +–V-+ | | || |DP11| | 1xx | || +—–+—-+——+ | | ++========================+ Call_Sent | | | +—-/\—-+——+ | | On 100,180,2xx process DP14 || | | | On 3xx, process DP12 || | | V On 486, process DP13 || | | +–+——-+ On 5xx, 6xx and 4xx || | | |Proceeding| (except 486) process DP21|| |

Gurbani, et al. Informational [Page 12] RFC 3976 Interworking SIP & IN January 2005

| +-+-+——+⇐========================++ | | | | | | | | | | | | | | | +–200——————+ | | +—-4xx to 6xx——–+ | | | | | +–V-+ | On DPs 21, 2, 4, 6, 8, 10 | | |DP14| | send 4xx-6xx final response | | +——–+—-+–+ +——-+ | | | O_Alerting |

       |                     |  |     +---------+------+
    +--V-------+             |  |               |
    |Completed |<------------+  |            +--V-+
    +--+-------+                |            |DP16|
       |                        |     +------+----+----+
    +--V-------+                |   +-+ O_Active       |
    |Terminated|<---------------+   | +-------------+--+
    +----------+                    |               |
                              +-----+            +--V-+
                              |                  |DP19|
                           +--V-+       +--------+----+
                           |DP17|       | O_Disconnect|
                           +--+-+       +-------------+
                              |
                              V
                         To O_EXCEPTION
    Legend:
    | Communication between
    | states in the same
    V protocol
    ======> Communication between IN Layer and SIP Protocol
            State machine to transfer call state
       Figure 5.  Mapping from SIP to O_BCSM
 The SIP "Calling" protocol state has enough functionality to absorb
 the seven PICs as described below:
    O_NULL: This PIC is basically a fall through state to the next
    PIC, AUTHORIZE_ORIGINATION_ATTEMPT.
    AUTHORIZE_ORIGINATION_ATTEMPT: In this PIC, the IN layer has
    detected that someone wishes to make a call.  Under some
    circumstances (e.g., if the user is not allowed to make calls
    during certain hours), such a call cannot be placed.  SIP can
    authorize the calling party by using a set of policy directives

Gurbani, et al. Informational [Page 13] RFC 3976 Interworking SIP & IN January 2005

    configured by the SIP administrator.  If the called party is
    authorized to place the call, the IN layer is instructed to enter
    the next PIC, COLLECT_INFO through DP 3
    (Origination_Attempt_Authorized).  If for some reason the call
    cannot be authorized, DP 2 (Origination_Denied) is processed, and
    control transfers to the SIP state machine.  The SIP state machine
    must format and send a non-2xx final response (possibly 403) to
    the upstream entity.
    COLLECT_INFO: This PIC is responsible for collecting a dial string
    from the calling party and verifying the format of the string.  If
    overlap dialing is being used, this PIC can invoke DP 4
    (Collect_Timeout) and transfer control to the SIP state machine,
    which will format and send a non-2xx final response (possibly a
    484).  If the dial string is valid, DP 5 (Collected_Info) is
    processed, and the IN layer is instructed to enter the next PIC,
    ANALYZE_INFO.
    ANALYZE_INFO: This PIC is responsible for translating the dial
    string to a routing number.  Many IN services, such as freephone,
    LNP (Local Number Portability), and OCS (Originating Call
    Screening) occur during this PIC.  The IN layer can use the R-URI
    of the SIP INVITE request for analysis.  If the analysis succeeds,
    the IN layer is instructed to enter the next PIC, SELECT_ROUTE.
    If the analysis fails, DP 6 (Invalid_Info) is processed, and the
    control transfers to the SIP state machine, which will generate a
    non-2xx final response (possibly 400, 401, 403, 404, 405, 406,
    410, 414, 415, 416, 485, or 488) and send it to the upstream
    entity.
    SELECT_ROUTE: In the circuit-switched network, the actual physical
    route has to be selected at this point.  The SIP analogue would be
    to determine the next hop SIP server.  This could be chosen by a
    variety of means.  For instance, if the Request URI in the
    incoming INVITE request is an E.164 number, the SIP entity can use
    a protocol like TRIP [10] to find the best gateway to egress the
    request onto the PSTN.  If a successful route is selected, the IN
    call model moves to PIC AUTH_CALL_SETUP via DP 9 (Route_Selected).
    Otherwise, the control transfers to the SIP state machine via DP 8
    (Route_Select_Failure), which will generate a non-2xx final
    response (possibly 488) and send it to the upstream entity.
    AUTH_CALL_SETUP: Certain service features restrict the type of
    call that may originate on a given line or trunk.  This PIC is the
    point at which relevant restrictions are examined.  If no such
    restrictions are encountered, the IN call model moves to PIC
    CALL_SENT via DP 11 (Origination_Authorized).  If a restriction is
    encountered that prohibits further processing of the call, DP 10

Gurbani, et al. Informational [Page 14] RFC 3976 Interworking SIP & IN January 2005

    (Authorization_Failure) is processed, and control is transferred
    to the SIP state machine, which will generate a non-2xx final
    response (possibly 404, 488, or 502).  Otherwise, DP 11
    (Origination_Authorized) is processed, and the IN layer is
    instructed to enter the next PIC, CALL_SENT.
    CALL_SENT: At this point, the request needs to be sent to the
    downstream entity.  The IN layer waits for a signal confirming
    either that the call has been presented to the called party or
    that a called party cannot be reached for a particular reason.
    The control is transferred to the SIP state machine.  The SIP
    state machine should now send the call to the next downstream
    server determined in PIC SELECT_ROUTE.  The IN call model now
    blocks until unblocked by the SIP state machine.
    If the above seven PICs have been successfully negotiated, the
    SIN-enabled SIP entity now sends the SIP INVITE message to the
    next hop server.  Further processing now depends on the
    provisional responses (if any) and the final response received by
    the SIP protocol state machine.  The core SIP specification does
    not guarantee the delivery of 1xx responses; thus special
    processing is needed at the IN layer to transition to the next PIC
    (O_ALERTING) from the CALL_SENT PIC.  The special processing
    needed for responses while the SIP state machine is in the
    "Proceeding" state and the IN layer is in the "CALL_SENT" state is
    described next.
       A 100 response received at the SIP state machine elicits no
       special behavior in the IN layer.
       A 180 response received at the SIP entity enables the
       processing of DP 14 (O_Term_Seized), however, a state
       transition to O_ALERTING is not undertaken yet.  Instead, the
       IN layer is instructed to remain in the CALL_SENT PIC until a
       final response is received.
       A 2xx response received at the SIP entity enables the
       processing of DP 14 (O_Term_Seized), and the immediate
       transition to the next state, O_ALERTING (processing in
       O_ALERTING is described later).
       A 3xx response received at the SIP entity enables the
       processing of DP 12 (Route_Failure).  The IN call model from
       this point goes back to the SELECT_ROUTE PIC to select a new
       route for the contacts in the 3xx final response (not shown in
       Figure 5 for brevity).

Gurbani, et al. Informational [Page 15] RFC 3976 Interworking SIP & IN January 2005

       A 486 (Busy Here) response received at the SIP entity enables
       the processing of DP 13 (O_Called_Party_Busy) and resources for
       the call are released at the IN call model.
       If the SIN-enabled SIP entity gets a 4xx (except 486), 5xx, or
       6xx final response, DP 21 (O_Calling_Party_Disconnect &
       O_Abandon) is processed and control passes to the SIP state
       machine.  Since a call was not successfully established, both
       the IN layer and the SIP state machine can release resources
       for the call.
    O_ALERTING - This PIC will be entered as a result of receiving a
    200-class response.  Since a 200-class response to an INVITE
    indicates acceptance, this PIC is mostly a fall through to the
    next PIC, O_ACTIVE via DP 16 (O_Answer).
    O_ACTIVE - At this point, the call is active.  Once in this state,
    the call may get disconnected only when one of the following three
    events occur: (1) the network connection fails, (2) the called
    party disconnects the call, or (3) the calling party disconnects
    the call.  If event (1) occurs, DP 17 (O_Connection_Failure) is
    processed and call control is transferred to the SIP protocol
    state machine.  Since the network failed, there is not much sense
    in attempting to send a BYE request; thus, both the SIP protocol
    state machine and the IN call layer should release all resources
    associated with the call and initialize themselves to the null
    state.  Event (2) results in the processing of DP 19
    (O_DISCONNECT) and a move to the last PIC, O_DISCONNECT.  Event
    (3) occurs if the calling party deliberately terminated the call.
    In this case, DP 21 (O_Abandon & O_Calling_Party_Disconnect) will
    be processed, and control will be passed to the SIP protocol state
    machine.  The SIP protocol state machine must send a BYE request
    and wait for a final response.  The IN layer releases all of its
    resources and initializes itself to the null state.
    O_DISCONNECT: When the SIP entity receives a BYE request, the IN
    layer is instructed to move to the last PIC, O_DISCONNECT via DP
    19.  A final response for the BYE is generated and transmitted by
    the SIP entity, and the call resources are freed by both the SIP
    protocol state machine and the IN layer.

5.2. Mapping SIP Protocol State Machine to T_BCSM

 The T_BCSM object is created when a SIP INVITE message makes its way
 to the terminating SIN-enabled SIP entity.  This entity creates the
 T_BCSM object and initializes it to the T_NULL PIC.

Gurbani, et al. Informational [Page 16] RFC 3976 Interworking SIP & IN January 2005

 Figure 6 provides a visual map from the SIP protocol state machine to
 the terminating half of the IN call model:
         SIP                                      T_BCSM
      | INVITE
      V
 +----------+                          +------------+
 |Proceeding+=========================>+ T_Null     +<-------+
 +-+--+--/\-+                          +/\----+-----+        |
   |  |  ||        +-----------+        |     |              |
   |  |  ||<=======+T_Exception+--------+  +--V-+         +--+-+
   |  |  ||        +-/\--------+           |DP22|         |DP35|
   |  |  ||          |    +----+       +---+----+------+  +--+-+
   |  |  ||          +<---+DP23|<------+Auth._Term._Att+---->+
   |  |  ||          |    +----+       +------+--------+     |
   |  |  ||          |                        |              |
   |  |  ||          |                     +--V-+            |
   |  |  ||          |                     |DP24|            |
   |  |  ||          |    +----+       +---+----+------+     |
   |  |  ||          +<---+DP25|<------+Select_Facility+---->+
   |  |  ||          |    +----+       +------+--------+     |
   |  |  ||          |                        |              |
   |  |  ||          |                     +--V-+            |
   |  |  ||          |                     |DP26|            |
   |  |  ||          |    +----+       +---+----+------+     |
   |  |  ||          +<---+DP27|<------+ Present_Call  +---->+
   |  |  ||          |    +----+       +------+--------+     |
   |  |  ||          |                        |              |
   |  |  ||          |                     +--V-+            |
   |  |  ||          |                     |DP28|            |
   |  |  ||          |    +----+       +---+----+------+     |
   |  |  ||          +<---+DP29|<------+ T_Alerting    +---->+
   |  |  ||          |    +----+       +-/\--+---------+     |
   |  |  ||          +<--------------+   ||   |              |
   |  |  ||                          |   ||   |              |
   |  |  ++==========================|===++   |              |
   |  |  /\                  +-------+     +--V-+            |
   |  |  ||                  |             +DP30|            |
   |  |  ||                +-+--+      +---+----+------+     |
   |  |  ||                |DP31+<-----| T_Active      +---->+
   |  |  ||                +----+      +-/\-----+------+
   |  |  ||                              ||      |
   |  |  ||                              ||      |

2xx | | ++==============================++ | sent | | | +—-+ | 3xx - 6xx response +–V-+

sent DP33

Gurbani, et al. Informational [Page 17] RFC 3976 Interworking SIP & IN January 2005

Completed T_Disconnect
Confirmed

+——>|

      |
 +----V-----+
 |Terminated|
 +----------+
   Legend:
   | Communication between
   | states in the same
   V protocol
   ======> Communication between IN call model and SIP
           protocol state machine to transfer call state
      Figure 6.  Mapping from SIP to T_BCSM
 The SIP "Proceeding" state has enough functionality to absorb the
 first five PICS -- T_Null, Authorize_Termination_Attempt,
 Select_Facility, Present_Call, T_Alerting -- as described below:
    T_NULL:  At this PIC, the terminating end creates the call at the
    IN layer.  The incoming call results in the processing of DP 22,
    Termination_Attempt, and a transition to the next PIC,
    AUTHORIZE_TERMINATION_ATTEMPT, takes place.
    AUTHORIZE_TERMINATION_ATTEMPT: At this PIC, it is ascertained that
    the called party wishes to receive the call and that the
    facilities of the called party are compatible with those of the
    calling party.  If any of these conditions is not met, DP 23
    (Termination_Denied) is invoked, and the call control is
    transferred to the SIP protocol state machine.  The SIP protocol
    state machine can format and send a non-2xx final response
    (possibly 403, 405, 415, or 480).  If the conditions of the PIC
    are met, processing of DP 24 (Termination_Authorized) is invoked,
    and a transition to the next PIC, SELECT_FACILITY, takes place.

Gurbani, et al. Informational [Page 18] RFC 3976 Interworking SIP & IN January 2005

    SELECT_FACILITY: In circuit switched networks, this PIC is
    intended to select a line or trunk to reach the called party.  As
    lines or trunks are not applicable in an IP network, a SIN-enabled
    SIP entity can use this PIC to interface with a PSTN gateway and
    select a line/trunk to route the call.  If the called party is
    busy, or if a line/trunk cannot be seized, the processing of DP 25
    (T_Called_Party_Busy) is invoked, and the call goes to the SIP
    protocol state machine.  The SIP protocol state machine must
    format and send a non-2xx final response (possibly 486 or 600).
    If a line/trunk was successfully seized, the processing of DP 26
    (Terminating_Resource_Available) is invoked, and a transition to
    the next PIC, PRESENT_CALL, takes place.
    PRESENT_CALL: At this point, the call is being presented (via the
    ISUP ACM message, or Q.931 Alerting message, or simply by ringing
    a POTS phone).  If there was an error presenting the call, the
    processing of DP 27 (Presentation_Failure) is invoked, and the
    call control is transferred to the SIP protocol state machine,
    which must format and send a non-2xx final response (possibly
    480).  If the call was successfully presented, the processing of
    DP 28 (T_Term_Seized) is invoked, and a transition to the next
    PIC, T_ALERTING, takes place.
    T_ALERTING: At this point, the called party is being "alerted".
    Control now passes momentarily to the SIP protocol state machine
    so that it can generate and send a "180 Ringing" response to its
    peer.  Furthermore, since network resources have been allocated
    for the call, timers are set to prevent indefinite holding of such
    resources.  The expiration of the relevant timers results in the
    processing of DP 29 (T_No_Answer), and the call control is
    transferred to the SIP protocol state machine, which must format
    and send a non-2xx final response (possibly 408).  If the called
    party answers, then DP 30 (T_Answer) is processed, followed by a
    transition to the next PIC, T_ACTIVE.
 After the above five PICs have been negotiated, the rest are mapped
 as follows:
    T_ACTIVE: The call is now active.  Once this state is reached, the
    call may become inactive under one of the following three
    conditions: (1) The network fails the connection, (2) the called
    party disconnects the call, or (3) the calling party disconnects
    the call.  Event (1) results in the processing of DP 31
    (T_Connection_Failure), and call control is transferred to the SIP
    protocol state machine.  Since the network failed, there is little
    sense in attempting to send a BYE request; thus, both the SIP
    protocol state machine and the IN call layer should release all
    resources associated with the call and initialize themselves to

Gurbani, et al. Informational [Page 19] RFC 3976 Interworking SIP & IN January 2005

    the null state.  Event (2) results in the processing of DP 33
    (T_Disconnect) and a transition to the next PIC, T_DISCONNECT.
    Event (3) occurs at the receipt of a BYE request at the SIP
    protocol state machine (not shown in Figure 6).  Resources for the
    call should be deallocated, and the SIP protocol state machine
    must send a 200 OK for the BYE request (not shown in Figure 6).
    T_DISCONNECT: In this PIC, the disconnect treatment associated
    with the called party's having disconnected the call is performed
    at the IN layer.  The SIP protocol state machine sends out a BYE
    and awaits a final response for the BYE (not shown in Figure 6).

6. Examples of Call Flows

 Two examples are provided here to show how SIP protocol state machine
 and the IN call model work synchronously with each other.
 In the first example, a SIP UAC originates a call request destined to
 an 800 freephone number:
    INVITE sip:18005551212@example.com SIP/2.0
    From: sip:16305551212@example.net;tag=991-7as-66ff
    To: sip:18005551212@example.com
    Via: SIP/2.0/UDP stn1.example.net
    Call-ID: 67188121@example.net
    CSeq: 1 INVITE
 The request makes its way to the originating SIP network server
 running an IN call model.  The SIP network server hands, at the very
 least, the To: field and the From: field to the IN layer for
 freephone number translation.  The IN layer proceeds through its PICs
 and at the ANALYSE_INFO PIC consults the SCP for freephone
 translation.  The translated number is returned to the SIP network
 server, which forwards the message to the next hop SIP proxy, with
 the freephone number replaced by the translated number:
    INVITE sip:18475551212@example.com SIP/2.0
    From: sip:16305551212@example.net;tag=991-7as-66ff
    To: sip:18005551212@example.com
    Via: SIP/2.0/UDP ext-stn2.example.net
    Via: SIP/2.0/UDP stn1.example.net
    Call-ID: 67188121@example.net
    CSeq: 1 INVITE

Gurbani, et al. Informational [Page 20] RFC 3976 Interworking SIP & IN January 2005

 In the next example, a SIP UAC originates a call request destined to
 a 900 number:
    INVITE sip:19005551212@example.com SIP/2.0
    From: sip:16305551212@example.net;tag=991-7as-66dd
    To: sip:19005551212@example.com
    Via: SIP/2.0/UDP stn1.example.net
    Call-ID: 88112@example.net
    CSeq: 1 INVITE
 The request makes its way to the originating SIP network server
 running an IN call model.  The SIP network server hands, at the very
 least, the To: field and the From: field to the IN layer for 900
 number translation.  The IN layer proceeds through its PICs and at
 the ANALYSE_INFO PIC consults the SCP for the translation.  During
 the translation, the SCP detects that the originating party is not
 allowed to make 900 calls.  It passes this information to the
 originating SIP network server, which informs the SIP UAC by using a
 SIP "403 Forbidden" response status code:
    SIP/2.0 403 Forbidden
    From: sip:16305551212@example.net;tag=991-7as-66dd
    To: sip:19005551212@example.com;tag=78K-909II
    Via: SIP/2.0/UDP stn1.example.net
    Call-ID: 88112@example.net
    CSeq: 1 INVITE

7. Security Considerations

 Security considerations for SIN services cover both networks being
 used, namely, the PSTN and the Internet.  SIN uses the security
 measures in place for both the networks.  With reference to Figure 2,
 the INAP messages between the SCP and the SIN-enabled SIP entity must
 be secured by the signaling transport used between the SCP and the
 SIN-enabled entity.  Likewise, the requests coming into the SIN-
 enabled SIP entity must first be authenticated and, if need be,
 encrypted as well, using the means and procedures defined in [3] for
 SIP requests.

8. References

8.1. Normative References

 [1]   I. Faynberg, L. Gabuzda, M. Kaplan, and N.Shah, "The
       Intelligent Network Standards: Their Application to Services,"
       McGraw-Hill, 1997.

Gurbani, et al. Informational [Page 21] RFC 3976 Interworking SIP & IN January 2005

 [2]   ITU-T Q.1204 1993: Recommendation Q.1204, "Intelligent Network
       Distributed Functional Plane Architecture," International
       Telecommunications Union Standardization Section, Geneva.
 [3]   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.

8.2. Informative References

 [4]   ITU-T Q.1208: "General aspects of the Intelligent Network
       Application protocol"
 [5]   Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.
 [6]   Roach, A.B., "Session Initiation Protocol (SIP)-Specific Event
       Notification", RFC 3265, June 2002.
 [7]   Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
       Telephony Tones and Telephony Signals", RFC 2833, May 2000.
 [8]   ITU-T Q.1218: "Interface Recommendation for Intelligent Network
       Capability Set 1".
 [9]   ITU-T Q.1228: "Interface Recommendation for Intelligent Network
       Capability Set 2".
 [10]  Rosenberg, J., Salama, H., and M. Squire, "Telephony Routing
       over IP (TRIP)", RFC 3219, January 2002.

Gurbani, et al. Informational [Page 22] RFC 3976 Interworking SIP & IN January 2005

Appendix A: Mapping of 4xx-6xx Responses in SIP to IN Detections Points

 The mapping of error codes 4xx-6xx responses in SIP to the possible
 Detection Points in PIC Originating and Terminating Call Handling is
 indicated in the table below.  The reason phrase in the 4xx-6xx
 response is reproduced from [3].
      SIP response code             DP mapping to IN
      -----------------             ----------------------
      200 OK                        DP 14
      3xx                           DP 12
      403 Forbidden                 DP 2,  DP 21
      484 Address Incomplete        DP 4,  DP 21
      400 Bad Request               DP 6,  DP 21
      401 Unauthorized              DP 6,  DP 21
      403 Forbidden                 DP 6,  DP 21, DP 23
      404 Not Found                 DP 6,  DP 21
      405 Method Not Allowed        DP 6,  DP 21, DP 23
      406 Not Acceptable            DP 6,  DP 21
      408 Request Timeout           DP 29
      410 Gone                      DP 6,  DP 21
      414 Request-URI Too Long      DP 6,  DP 21
      415 Unsupported Media Type    DP 6,  DP 21, DP 23
      416 Unsupported URI Scheme    DP 6,  DP 21
      480 Temporarily Unavailable   DP 23, DP 27
      485 Ambiguous                 DP 6,  DP 21
      486 Busy Here                 DP 13, DP 21, DP 25
      488 Not Acceptable Here       DP 6,  DP 21

Gurbani, et al. Informational [Page 23] RFC 3976 Interworking SIP & IN January 2005

Acknowledgments

 Special acknowledgment is due to Hui-Lan Lu for acting as the chair
 of the SIN DT and ensuring that the focus of the DT did not veer too
 far.  The authors would also like to give special thanks to Mr. Ray
 C. Forbes from Marconi Communications Limited for his valuable
 contribution on the system and network architectural aspects as co-
 chair in the ETSI SPAN.   Thanks also to Doris Lebovits, Kamlesh
 Tewani, Janusz Dobrowloski, Jack Kozik, Warren Montgomery, Lev
 Slutsman, Henning Schulzrinne, and Jonathan Rosenberg, who all
 contributed to the discussions on the relationship of IN and SIP call
 models.

Author's Addresses

 Vijay K. Gurbani
 Lucent Technologies, Inc.
 2000 Lucent Lane, Rm 6G-440
 Naperville, Illinois 60566
 USA
 Phone: +1 630 224 0216
 EMail: vkg@lucent.com
 Frans Haerens
 Alcatel Bell
 Francis Welles Plein,1
 Belgium
 Phone: +32 3 240 9034
 EMail: frans.haerens@alcatel.be
 Vidhi Rastogi
 Wipro Technologies
 Plot No.72, Keonics Electronics City,
 Hosur Main Road,
 Bangalore 226 560 100
 Phone: +91 80 51381869
 EMail: vidhi.rastogi@wipro.com

Gurbani, et al. Informational [Page 24] RFC 3976 Interworking SIP & IN January 2005

Full Copyright Statement

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 contained in BCP 78 and at www.rfc-editor.org, and except as set
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Acknowledgement

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Gurbani, et al. Informational [Page 25]

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