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RFC 3372 - Session Initiation Protocol for Telephones (SIP-T): C

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Network Working Group                                          A. Vemuri
Request for Comments: 3372                          Qwest Communications
BCP: 63                                                      J. Peterson
Category: Best Current Practice                                  NeuStar
                                                          September 2002

          Session Initiation Protocol for Telephones (SIP-T):
                       Context and Architectures

Status of this Memo

   This document specifies an Internet Best Current Practices for the
   Internet Community, and requests discussion and suggestions for
   improvements.  Distribution of this memo is unlimited.

Copyright Notice



   The popularity of gateways that interwork between the PSTN (Public
   Switched Telephone Network) and SIP networks has motivated the
   publication of a set of common practices that can assure consistent
   behavior across implementations.  This document taxonomizes the uses
   of PSTN-SIP gateways, provides uses cases, and identifies mechanisms
   necessary for interworking.  The mechanisms detail how SIP provides
   for both 'encapsulation' (bridging the PSTN signaling across a SIP
   network) and 'translation' (gatewaying).

Table of Contents

   2.  SIP-T for ISUP-SIP Interconnections  . . . . . . . . . . . . .  4
   3.  SIP-T Flows  . . . . . . . . . . . . . . . . . . . . . . . . .  7
   3.1 SIP Bridging (PSTN - IP - PSTN)  . . . . . . . . . . . . . . .  8
   3.2 PSTN origination - IP termination  . . . . . . . . . . . . . .  9
   3.3 IP origination - PSTN termination  . . . . . . . . . . . . . . 11
   4.  SIP-T Roles and Behavior . . . . . . . . . . . . . . . . . . . 12
   4.4 Behavioral Requirements Summary  . . . . . . . . . . . . . . . 15
   5.  Components of the SIP-T Protocol . . . . . . . . . . . . . . . 16

   5.4 Support for mid-call signaling . . . . . . . . . . . . . . . . 17
   6.  SIP Content Negotiation  . . . . . . . . . . . . . . . . . . . 17

1. Introduction

   The Session Initiation Protocol (SIP [1]) is an application-layer
   control protocol that can establish, modify and terminate multimedia
   sessions or calls.  These multimedia sessions include multimedia
   conferences, Internet telephony and similar applications.  SIP is one
   of the key protocols used to implement Voice over IP (VoIP).
   Although performing telephony call signaling and transporting the
   associated audio media over IP yields significant advantages over
   traditional telephony, a VoIP network cannot exist in isolation from
   traditional telephone networks.  It is vital for a SIP telephony
   network to interwork with the PSTN.

   The popularity of gateways that interwork between the PSTN and SIP
   networks has motivated the publication of a set of common practices
   that can assure consistent behavior across implementations.  The
   scarcity of SIP expertise outside the IETF suggests that the IETF is
   the best place to stage this work, especially since SIP is in a
   relative state of flux compared to the core protocols of the PSTN.
   Moreover, the IETF working groups that focus on SIP (SIP and SIPPING)
   are best positioned to ascertain whether or not any new extensions to
   SIP are justified for PSTN interworking.  This framework addresses
   the overall context in which PSTN-SIP interworking gateways might be
   deployed, provides use cases and identifies the mechanisms necessary
   for interworking.

   An important characteristic of any SIP telephony network is feature
   transparency with respect to the PSTN.  Traditional telecom services
   such as call waiting, freephone numbers, etc., implemented in PSTN
   protocols such as Signaling System No. 7 (SS7 [6]) should be offered
   by a SIP network in a manner that precludes any debilitating
   difference in user experience while not limiting the flexibility of
   SIP.  On the one hand, it is necessary that SIP support the
   primitives for the delivery of such services where the terminating
   point is a regular SIP phone (see definition in Section 2 below)
   rather than a device that is fluent in SS7.  However, it is also
   essential that SS7 information be available at gateways, the points

   of SS7-SIP interconnection, to ensure transparency of features not
   otherwise supported in SIP.  If possible, SS7 information should be
   available in its entirety and without any loss to trusted parties in
   the SIP network across the PSTN-IP interface; one compelling need to
   do so also arises from the fact that certain networks utilize
   proprietary SS7 parameters to transmit certain information through
   their networks.

   Another important characteristic of a SIP telephony network is
   routability of SIP requests - a SIP request that sets up a telephone
   call should contain sufficient information in its headers to enable
   it to be appropriately routed to its destination by proxy servers in
   the SIP network.  Most commonly this entails that parameters of a
   call like the dialed number should be carried over from SS7 signaling
   to SIP requests.  Routing in a SIP network may in turn be influenced
   by mechanisms such as TRIP [8] or ENUM [7].

   The SIP-T (SIP for Telephones) effort provides a framework for the
   integration of legacy telephony signaling into SIP messages.  SIP-T
   provides the above two characteristics through techniques known as
   'encapsulation' and 'translation' respectively.  At a SIP-ISUP
   gateway, SS7 ISUP messages are encapsulated within SIP in order that
   information necessary for services is not discarded in the SIP
   request.  However, intermediaries like proxy servers that make
   routing decisions for SIP requests cannot be expected to understand
   ISUP, so simultaneously, some critical information is translated from
   an ISUP message into the corresponding SIP headers in order to
   determine how the SIP request will be routed.

   While pure SIP has all the requisite instruments for the
   establishment and termination of calls, it does not have any baseline
   mechanism to carry any mid-call information (such as the ISUP INF/INR
   query) along the SIP signaling path during the session.  This mid-
   call information does not result in any change in the state of SIP
   calls or the parameters of the sessions that SIP initiates.  A
   provision to transmit such optional application-layer information is
   also needed.

   Problem definition: To provide ISUP transparency across SS7-SIP

   SS7-SIP Interworking Requirements     SIP-T Functions
   Transparency of ISUP                  Encapsulation of ISUP in the
   Signaling                             SIP body

   Routability of SIP messages with      Translation of ISUP information
   dependencies on ISUP                  into the SIP header

   Transfer of mid-call ISUP signaling   Use of the INFO Method for mid-
   messages                              call signaling

   Table 1: SIP-T features that fulfill PSTN-IP inter-connection

   While this document specifies the requirements above, it provide
   mechanisms to satisfy them - however, this document does serve as an
   framework for the documents that do provide these mechanisms, all of
   which are referenced in Section 5.

   Note that many modes of signaling are used in telephony (SS7 ISUP,
   BTNUP, Q.931, MF etc.).  This document focuses on SS7 ISUP and aims
   to specify the behavior across ISUP-SIP interfaces only.  The scope
   of the SIP-T enterprise may, over time, come to encompass other
   signaling systems as well.

2. SIP-T for ISUP-SIP Interconnections

   SIP-T is not a new protocol - it is a set of mechanisms for
   interfacing traditional telephone signaling with SIP.  The purpose of
   SIP-T is to provide protocol translation and feature transparency
   across points of PSTN-SIP interconnection.  It intended for use where
   a VoIP network (a SIP network, for the purposes of this document)
   interfaces with the PSTN.

   Using SIP-T, there are three basic models for how calls interact with
   gateways.  Calls that originate in the PSTN can traverse a gateway to
   terminate at a SIP endpoint, such as an IP phone.  Conversely, an IP
   phone can make a call that traverses a gateway to terminate in the
   PSTN.  Finally, an IP network using SIP may serve as a transit
   network between gateways - a call may originate and terminate in the
   PSTN, but cross a SIP-based network somewhere in the middle.

   The SS7 interfaces of a particular gateway determine the ISUP
   variants that that gateway supports.  Whether or nor a gateway
   supports a particular version of ISUP determines whether it can
   provide feature transparency while terminating a call.

   The following are the primary agents in a SIP-T-enabled network.

   o  PSTN (Public Switched Telephone Network): This refers to the
      entire interconnected collection of local, long-distance and
      international phone companies.  In the examples below, the term
      Local Exchange Carrier (LEC) is used to denote a portion (usually,
      a regional division) of the PSTN.

   o  IP endpoints: Any SIP user agent that can act as an originator or
      recipient of calls.  Thus, the following devices are classified as
      IP endpoints:

      *  Gateways: A telephony gateway provides a point of conversion
         between signaling protocols (such as ISUP and SIP) as well as
         circuit-switch and packet-switched audio media.  The term Media
         Gateway Controller (MGC) is also used in the examples and
         diagrams in this document to denote large-scale clusters of
         decomposed gateways and control logic that are frequently
         deployed today.  So for example, a SIP-ISUP gateway speaks ISUP
         to the PSTN and SIP to the Internet and is responsible for
         converting between the types of signaling, as well as
         interchanging any associated bearer audio media.

      *  SIP phones: The term used to represent all end-user devices
         that originate or terminate SIP VoIP calls.

      *  Interface points between networks where administrative policies
         are enforced (potentially middleboxes, proxy servers, or

   o  Proxy Servers: A proxy server is a SIP intermediary that routes
      SIP requests to their destinations.  For example, a proxy server
      might direct a SIP request to another proxy, a gateway or a SIP

                        ***                    ***
                       *                         *
                      *    -------                *
                     *     |proxy|                 *
                    *      -------                  *
                |----|                            |----|
               /|MGC1|       VoIP Network         |MGC2|\
              /  ----                              ----  \
      SS7    /       *                               *    \ SS7
            /         *           -------           *      \
           /           *          |proxy|          *        \
       | LEC1 |          **                     **      | LEC2  |

   Figure 1: Motivation for SIP-T in ISUP-SIP interconnection

   In Figure 2 a VoIP cloud serves as a transit network for telephone
   calls originating in a pair of LECs, where SIP is employed as the
   VoIP protocol used to set up and tear down these VoIP calls.  At the
   edge of the depicted network, an MGC converts the ISUP signals to SIP
   requests,  and sends them to a proxy server which in turn routes
   calls on other MGCs.  Although this figure depicts only two MGCs,
   VoIP deployments would commonly have many such points of
   interconnection with the PSTN (usually to diversify among PSTN rate
   centers).  For a call originating from LEC1 and be terminating in
   LEC2, the originator in SIP-T is the gateway that generates the SIP
   request for a VoIP call, and the terminator is the gateway that is
   the consumer of the SIP request; MGC1 would thus be the originator
   and MGC2, the terminator.  Note that one or more proxies may be used
   to route the call from the originator to the terminator.

   In this flow, in order to seamlessly integrate the IP network with
   the PSTN, it is important to preserve the received SS7 information
   within SIP requests at the originating gateway and reuse this SS7
   information when signaling to the PSTN at the terminating gateway.
   By encapsulating ISUP information in the SIP signaling, a SIP network
   can ensure that no SS7 information that is critical to the
   instantiation of features is lost when SIP bridges calls between two
   segments of the PSTN.

   That much said, if only the exchange of ISUP between gateways were
   relevant here, any protocol for the transport of signaling
   information may be used to achieve this, obviating the need for SIP
   and consequently that of SIP-T.  SIP-T is employed in order to
   leverage the intrinsic benefits of utilizing SIP: request routing and
   call control leveraging proxy servers (including the use of forking),

   ease of SIP service creation, SIP's capability negotiation systems,
   and so on.  Translation of information from the received ISUP message
   parameters to SIP header fields enables SIP intermediaries to
   consider this information as they handle requests.  SIP-T thus
   facilitates call establishment and the enabling of new telephony
   services over the IP network while simultaneously providing a method
   of feature-rich interconnection with the PSTN.

   Finally, the scenario in Figure 2 is just one of several flows in
   which SIP-T can be used - voice calls do not always both originate
   and terminate in the PSTN (via gateways); SIP phones can also be
   endpoints in a SIP-T session.  In subsequent sections, the following
   possible flows will be further detailed:

   1.  PSTN origination - PSTN termination: The originating gateway
       receives ISUP from the PSTN and it preserves this information
       (via encapsulation and translation) in the SIP messages that it
       transmits towards the terminating gateway.  The terminator
       extracts the ISUP content from the SIP message that it receives
       and it reuses this information in signaling sent to the PSTN.

   2.  PSTN origination - IP termination: The originating gateway
       receives ISUP from the PSTN and it preserves this ISUP
       information in the SIP messages (via encapsulation and
       translation) that it directs towards the terminating SIP user
       agent.  The terminator has no use for the encapsulated ISUP and
       ignores it.

   3.  IP origination - PSTN termination: A SIP phone originates a VoIP
       call that is routed by one or more proxy servers to the
       appropriate terminating gateway.  The terminating gateway
       converts to ISUP signaling and directs the call to an appropriate
       PSTN interface, based on information that is present in the
       received SIP header.

   4.  IP origination - IP termination: This is a case for pure SIP.
       SIP-T (either encapsulation or translation of ISUP) does not come
       into play as there is no PSTN interworking.

3. SIP-T Flows

   The follow sections explore the essential SIP-T flows in detail.
   Note that because proxy servers are usually responsible for routing
   SIP requests (based on the Request-URI) the eventual endpoints at
   which a SIP request will terminate is generally not known to the
   originator.  So the originator does not select from the flows

   described in this section, as a matter of static configuration or on
   a per-call basis - rather, each call is routed by the SIP network
   independently, and it may instantiate any of the flows below as the
   routing logic of the network dictates.

3.1 SIP Bridging (PSTN - IP - PSTN)

                      ***                    ***
                     *                         *
                    *    -------                *
                   *     |proxy|                 *
                  *      -------                  *
               |---|                             |---|
              /|MGC|       VoIP Network          |MGC|\
             /  ---                               ---  \
            /     *                               *     \
           /       *            -------           *      \
          /          *          |proxy|          *        \
      | PSTN |          ***                    ***      | PSTN  |

   Figure 2: PSTN origination - PSTN termination (SIP Bridging)

   A scenario in which a SIP network connects two segments of the PSTN
   is referred to as 'SIP bridging'.  When a call destined for the SIP
   network originates in the PSTN, an SS7 ISUP message will eventually
   be received by the gateway that is the point of interconnection with
   the PSTN network.  This gateway is from the perspective of the SIP
   protocol the user agent client for this call setup request.
   Traditional SIP routing is used in the IP network to determine the
   appropriate point of termination (in this instance a gateway) and to
   establish a SIP dialog and begin negotiation of a media session
   between the origination and termination endpoints.  The egress
   gateway then signals ISUP to the PSTN, reusing any encapsulated ISUP
   present in the SIP request it receives as appropriate.

   A very elementary call-flow for SIP bridging is shown below.

       PSTN            MGC#1   Proxy    MGC#2          PSTN
       |-------IAM------>|       |        |              |
       |                 |-----INVITE---->|              |
       |                 |       |        |-----IAM----->|
       |                 |<--100 TRYING---|              |
       |                 |       |        |<----ACM------|
       |                 |<-----18x-------|              |
       |<------ACM-------|       |        |              |
       |                 |       |        |<----ANM------|
       |                 |<----200 OK-----|              |
       |<------ANM-------|       |        |              |
       |                 |------ACK------>|              |
       |-------REL------>|       |        |              |
       |<------RLC-------|------BYE------>|              |
       |                 |       |        |-----REL----->|
       |                 |       |        |<----RLC------|
       |                 |       |        |              |

3.2 PSTN origination - IP termination

                        ***                    ***
                       *                         *
                      *                           *
                     *                             *
                    *                               *
               /|MGC |       VoIP Network         |proxy|\
              /  ----                              -----  \
             /       *                               *     \
            /         *                             *       \
           /           *                           *         \
      | PSTN |          **                     **      | SIP phone |

   Figure 3: PSTN origination - IP termination

   A call originates from the PSTN and terminates at a SIP phone.  Note
   that in Figure 5, the proxy server acts as the registrar for the SIP
   phone in question.

   A simple call-flow depicting the ISUP and SIP signaling for a PSTN-
   originated call terminating at a SIP endpoint follows:

   PSTN           MGC                  Proxy              SIP phone
     |----IAM----->|                     |                     |
     |             |--------INVITE------>|                     |
     |             |                     |-------INVITE------->|
     |             |<------100 TRYING----|                     |
     |             |                     |<-------18x----------|
     |             |<---------18x--------|                     |
     |<----ACM-----|                     |                     |
     |             |                     |<-------200 OK-------|
     |<----ANM-----|                     |                     |
     |             |---------ACK-------->|                     |
     |-----REL---->|                     |                     |
     |             |----------BYE------->|                     |
     |<----RLC-----|                     |---------BYE-------->|
     |             |                     |                     |

3.3 IP origination - PSTN termination

                        ***                    ***
                       *                         *
                      *                           *
                     *                             *
                    *                               *
              /|proxy|       VoIP Network         |MGC |\
             /  -----                              ----  \
            /       *                               *     \
           /         *                             *       \
          /           *                           *         \
      |SIP phone |      **                     **      | PSTN  |

   Figure 4: IP origination - PSTN termination

   A call originates from a SIP phone and terminates in the PSTN.
   Unlike the previous two flows, there is therefore no ISUP
   encapsulation in the request - the terminating gateway therefore only
   performs translation on the SIP headers to derive values for ISUP

   A simple call-flow illustrating the different legs in the call is as
   shown below.

        SIP phone         Proxy                    MGC          PSTN
     |-----INVITE----->|                       |             |
     |                 |--------INVITE-------->|             |
     |<---100 TRYING---|                       |-----IAM---->|
     |                 |<------100 TRYING------|             |
     |<------18x-------|                       |             |
     |                 |<--------200 OK--------|             |
     |<-----200 OK-----|                       |             |
     |-------BYE------>|                       |             |
     |                 |----------BYE--------->|             |
     |                 |                       |-----REL---->|
     |<-----200 OK-----|                       |<----RLC-----|

4. SIP-T Roles and Behavior

   There are three distinct sorts of elements (from a functional point
   of view) in a SIP VoIP network that interconnects with the PSTN:

   1.  The originators of SIP signaling

   2.  The terminators of SIP signaling

   3.  The intermediaries that route SIP requests from the originator to
       the terminator

   Behavior for the Section 4.1, Section 4.2 and Section 4.3
   intermediary roles in a SIP-T call are described in the following

4.1 Originator

   The function of the originating user agent client is to generate the
   SIP Call setup requests (i.e., INVITEs).  When a call originates in
   the PSTN, a gateway is the UAC; otherwise some native SIP endpoint is
   the UAC.  In either case, note that the originator generally cannot
   anticipate what sort of entity the terminator will be, i.e., whether
   final destination of the request is in a SIP network or the PSTN.

   In the case of calls originating in the PSTN (see Figure 3 and Figure
   5), the originating gateway takes the necessary steps to preserve the
   ISUP information by encapsulating it in the SIP request it creates.
   The originating gateway is entrusted with the responsibility of
   identifying the version of the ISUP (ETSI, ANSI, etc.) that it has
   received and providing this information in the encapsulated ISUP
   (usually by adding a multipart MIME body with appropriate MIME
   headers).  It then formulates the headers of the SIP INVITE request
   from the parameters of the ISUP that it has received from the PSTN as
   appropriate (see Section 5).  This might, for instance, entail
   setting the 'To:' header field in the INVITE to the reflect dialed
   number (Called Party Number) of the received ISUP IAM.

   In other cases (like Figure 7), a SIP phone is the originator of a
   VoIP call.  Usually, the SIP phone sends requests to a SIP proxy that
   is responsible for routing the request to an appropriate destination.
   There is no ISUP to encapsulate at the user agent client, as there is
   no PSTN interface.  Although the call may terminate in the telephone
   network and need to signal ISUP in order for that to take place, the
   originator has no way to anticipate this and it would be foolhardy to
   require that all SIP VoIP user agents have the capability to generate
   ISUP.  It is therefore not the responsibility of an IP endpoints like
   a SIP phone to generate encapsulated ISUP.  Thus, an originator must
   generate the SIP signaling while performing ISUP encapsulation and
   translation when possible (meaning when the call has originated in
   the PSTN).

   Originator requirements: encapsulate ISUP, translate information from
   ISUP to SIP, multipart MIME support (for gateways only)

4.2 Terminator

   The SIP-T terminator is a consumer of the SIP calls.  The terminator
   is a standard SIP UA that can be either a gateway that interworks
   with the PSTN or a SIP phone.

   In case of PSTN terminations (see Figure 3 and Figure 7) the egress
   gateway terminates the call to its PSTN interface.  The terminator
   generates the ISUP appropriate for signaling to the PSTN from the
   incoming SIP message.  Values for certain ISUP parameters may be
   gleaned from the SIP headers or extracted directly from an
   encapsulated ISUP body.  Generally speaking, a gateway uses any
   encapsulated ISUP as a template for the message it will send, but it
   overwrites parameter values in the template as it translates SIP
   headers or adds any parameter values that reflect its local policies
   (see Appendix A item 1).

   In case of an IP termination (Figure 5), the SIP UAS that receives
   SIP messages with encapsulated ISUP typically disregards the ISUP
   message.  This does introduce a general requirement, however, that
   devices like SIP phones handle multipart MIME messages and unknown
   MIME types gracefully (this is a baseline SIP requirement, but also a
   place where vendors have been known to make shortcuts).

   Terminator requirements: standard SIP processing, interpretation of
   encapsulated ISUP (for gateways only), support for multipart MIME,
   graceful handling of unknown MIME content (for non-gateways only)

4.3 Intermediary

   Intermediaries like proxy servers are entrusted with the task of
   routing messages to one another, as well as gateways and SIP phones.
   Each proxy server makes a forwarding decision for a SIP request based
   on values of various headers, or 'routable elements' (including the
   Request-URI, route headers, and potentially many other elements of a
   SIP request).

   SIP-T does introduce some additional considerations for forwarding a
   request that could lead to new features and requirements for
   intermediaries.  Feature transparency of ISUP is central to the
   notion of SIP-T.  Compatibility between the ISUP variants of the
   originating and terminating PSTN interfaces automatically leads to
   feature transparency.  Thus, proxy servers might take an interest in
   the variants of ISUP that are encapsulated with requests - the
   variant itself could become a routable element.  The termination of a
   call at a point that results in greater proximity to the final
   destination (rate considerations) is also an important consideration.
   The preference of one over the other results in a trade-off between
   simplicity of operation and cost.  The requirement of procuring a
   reasonable rate may dictate that a SIP-T call spans dissimilar PSTN
   interfaces (SIP bridging across different gateways that don't support
   any ISUP variants in common).  In order to optimize for maximum
   feature transparency and rate, some operators of intermediaries might
   want to consider practices along the following lines:

   a) The need for ISUP feature transparency may necessitate ISUP
      variant translation (conversion), i.e., conversion from one
      variant of ISUP to another in order to facilitate the termination
      of that call over a gateway interface that does not support the
      ISUP variant of the originating PSTN interface.  (See Appendix A
      item 2.) Although in theory conversion may be performed at any
      point in the path of the request, it is optimal to perform it at a
      point that is at the greatest proximity to the terminating
      gateway.  This could be accomplished by delivering the call to an
      application that might perform the conversion between variants.
      Feature transparency in this case is contingent on the
      availability of resources to perform ISUP conversion, and it
      incurs an increase in the call-set up time.

   b) An alternative would be to sacrifice ISUP transparency by handing
      the call off to a gateway that does not support the version of the
      originating ISUP.  The terminating MGC would then just ignore the
      encapsulated ISUP and use the information in the SIP header to
      terminate the call.

   So, it may be desirable for proxy servers to have the intelligence to
   make a judicious choice given the options available to it.

   Proxy requirements: ability to route based on choice of routable

4.4 Behavioral Requirements Summary

   If the SIP-T originator is a gateway that received an ISUP request,
   it must always perform both encapsulation and translation ISUP,
   regardless of where the originator might guess that the request will

   If the terminator does not understand ISUP, it ignores it while
   performing standard SIP processing.  If the terminator does
   understand ISUP, and needs to signal to the PSTN, it should reuse the
   encapsulated ISUP if it understands the variant.  The terminator
   should perform the following steps:

   o  Extract the ISUP from the message body, and use this ISUP as a
      message template.  Note that if there is no encapsulated ISUP in
      the message, the gateway should use a canonical template for the
      message type in question (a pre-populated ISUP message configured
      in the gateway) instead.

   o  Translate the headers of the SIP request into ISUP parameters,
      overwriting any values in the message template.

   o  Apply any local policies in populating parameters.

   An intermediary must be able to route a call based on the choice of
   routable elements in the SIP headers.

5. Components of the SIP-T Protocol

   The mechanisms described in the following sections are the components
   of SIP-T that provide the protocol functions entailed by the

5.1 Core SIP

   SIP-T uses the methods and procedures of SIP as defined by RFC 3261.

5.2 Encapsulation

   Encapsulation of the PSTN signaling is one of the major requirements
   of SIP-T.  SIP-T uses multipart MIME bodies to enable SIP messages to
   contain multiple payloads (Session Description Protocol or SDP [5],
   ISUP, etc.).  Numerous ISUP variants are in existence today; the ISUP
   MIME type enable recipients too recognize the ISUP type (and thus
   determine whether or not they support the variant) in the most
   expeditious possible manner.  One scheme for performing ISUP
   encapsulation using multi-part MIME has been described in [2].

5.3 Translation

   Translation encompasses all aspects of signaling protocol conversion
   between SIP and ISUP.  There are essentially two components to the
   problem of translation:

   1.  ISUP SIP message mapping:  This describes a mapping between ISUP
       and SIP at the message level.  In SIP-T deployments gateways are
       entrusted with the task of generating a specific ISUP message for
       each SIP message received and vice versa.  It is necessary to
       specify the rules that govern the mapping between ISUP and SIP
       messages (i.e., what ISUP messages is sent when a particular SIP
       message is received: an IAM must be sent on receipt of an INVITE,
       a REL for BYE, and so on).  A potential mapping between ISUP and
       SIP messages has been described in [10].

   2.  ISUP parameter-SIP header mapping:  A SIP request that is used to
       set up a telephone call should contain information that enables
       it to be appropriately routed to its destination by proxy servers
       in the SIP network - for example, the telephone number dialed by
       the originating user.  It is important to standardize a set of
       practices that defines the procedure for translation of
       information from ISUP to SIP (for example, the Called Party
       Number in an ISUP IAM must be mapped onto the SIP 'To' header
       field and Request-URI, etc.).  This issue becomes inherently more
       complicated by virtue of the fact that the headers of a SIP
       request (especially an INVITE) may be transformed by
       intermediaries, and that consequently, the SIP headers and
       encapsulated ISUP bodies come to express conflicting values -
       effectively, a part of the encapsulated ISUP may be rendered
       irrelevant and obsolete.

5.4 Support for mid-call signaling

   Pure SIP does not have any provision for carrying any mid-call
   control information that is generated during a session.  The INFO [3]
   method should be used for this purpose.  Note however that INFO is
   not suitable for managing overlap dialing (for one way of
   implementing overlap dialing see [11]).  Also note that the use of
   INFO for signaling mid-call DTMF signals is not recommended (see
   RFC2833 [9] for a recommended mechanism).

6. SIP Content Negotiation

   The originator of a SIP-T request might package both SDP and ISUP
   elements into the same SIP message by using the MIME multipart
   format.  Traditionally in SIP, if the terminating device does not
   support a multipart payload (multipart/mixed) and/or the ISUP MIME
   type, it would then reject the SIP request with a 415 Unsupported
   Media Type specifying the media types it supports (by default,
   'application/SDP').  The originator would subsequently have to re-
   send the SIP request after stripping out the ISUP payload (i.e.  with
   only the SDP payload) and this would then be accepted.

   This is a rather cumbersome flow, and it is thus highly desirable to
   have a mechanism by which the originator could signify which bodies
   are required and which are optional so that the terminator can
   silently discard optional bodies that it does not understand
   (allowing a SIP phone to ignore an ISUP payload when processing ISUP
   is not critical).  This is contingent upon the terminator having
   support for a Content-type of multipart/mixed and access to the
   Content-Disposition header to express criticality.

   1.  Support for ISUP is optional.  Therefore, UA2 accepts the INVITE
       irrespective of whether it can process the ISUP.

   UA1                    UA2
      Content-type: application/sdp;
      Content-disposition: session; handling=required;
      Content-type: application/isup;
      Content-disposition: signal; handling=optional;)


   2.  Support for ISUP is preferred.  UA2 does not support the ISUP and
       rejects the INVITE with a 415 Unsupported Media Type.  UA1 strips
       off the ISUP and re-sends the INVITE with SDP only and this is
       the accepted.

   UA1                    UA2
   INVITE--> (Content-type:multipart/mixed;
      Content-type: application/sdp;
      Content-disposition: session; handling=required;
      Content-type: application/isup;
      Content-disposition: signal; handling=required;)

                     (Accept: application/sdp)


   (Content-type: application/sdp)


   3.  Support for ISUP is mandatory for call establishment.  UA2 does
       not support the ISUP and rejects the INVITE with a 415
       Unsupported Media type.  UA1 then directs its request to UA3.

   UA1                    UA2
   INVITE--> (Content-type:multipart/mixed;
      Content-type: application/sdp;
      Content-disposition: session; handling=required;
      Content-type: application/isup;
      Content-disposition: signal; handling=required;)

                  (Accept: application/sdp)


   UA1                   UA3
   INVITE--> (Content-type:multipart/mixed;
       Content-type: application/sdp;
       Content-disposition: session; handling=required;
       Content-type: application/isup;
       Content-disposition: signal; handling=required;)

   Note that the exchanges of messages above are not complete; only the
   messages relevant to this discussion are shown.  Specifics of the
   ISUP MIME type can be obtained from [2].  The 'version' and 'base'
   parameters are not shown here, but must be used in accordance with
   the rules of [2].

7. Security Considerations

   SIP-T can be employed as an interdomain signaling mechanism that may
   be subject to pre-existing trust relationships between administrative
   domains.  In many legal environments, distribution of ISUP is
   restricted to licensed carriers; SIP-T introduces some challenges in
   so far as it bridges carrier signaling with end-user signaling.  Any
   administrative domain implementing SIP-T should have an adequate
   security apparatus (including elements that manage any appropriate
   policies to manage fraud and billing in an interdomain environment)
   in place to ensure that the transmission of ISUP information does not
   result in any security violations.

   Transporting ISUP in SIP bodies may provide opportunities for abuse,
   fraud, and privacy concerns, especially when SIP-T requests can be
   generated, inspected or modified by arbitrary SIP endpoints.  ISUP
   MIME bodies should be secured (preferably with S/MIME [4]) to
   alleviate this concern, as is described in the Security
   Considerations of the core SIP specification [1].  Authentication
   properties provided by S/MIME would allow the recipient of a SIP-T
   message to ensure that the ISUP MIME body was generated by an

   authorized entity.  Encryption would ensure that only carriers
   possessing a particular decryption key are capable of inspecting
   encapsulated ISUP MIME bodies in a SIP request.

   SIP-T endpoints MUST support S/MIME signatures (CMS SignedData), and
   SHOULD support encryption (CMS EnvelopedData).

8. IANA Considerations

   This document introduces no new considerations for IANA.

Normative References

         Session Initiation Protocol", RFC 3261, May 2002.

   [2]   Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F.,
         Watson, M. and M. Zonoun, "MIME media types for ISUP and QSIG
         objects", RFC 3204, December 2001.

   [3]   Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.

   [4]   Ramsdell, B., "S/MIME Version 3 Message Specification", RFC
         2633, June 1999.

         Protocol", RFC 2327, April 1998.

Non-Normative References

   [6]   International Telecommunications Union, "Signaling System No.
         7; ISDN User Part Signaling procedures", ITU-T Q.764, September
         1997, <http://www.itu.int>.

   [7]   Faltstrom, P., "E.164 number and DNS", RFC 2916, September

   [8]   Rosenberg, J., Salama, H. and M. Squire, "Telephony Routing

   [9]   Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
         Telephony Tones and Telephony Signals", RFC 2833, May 2000.

   [10]  Camarillo, G., Roach, A., Peterson, J. and L. Ong, "ISUP to SIP
         Mapping",  Work in Progress.

   [11]  Camarillo, G., Roach, A., Peterson, J. and L. Ong, "Mapping of
         ISUP Overlap Signaling to SIP", Work in Progress.

Appendix A. Notes

   1.  Some terminating MGCs may alter the encapsulated ISUP in order to
       remove any conditions specific to the originating circuit; for
       example, continuity test flags in the Nature of Connection
       Indicators, etc.

   2.  Even so, the relevance of ANSI-specific information in an ETSI
       network (or vice versa) is questionable.  Clearly, the strength
       of SIP-T is realized when the encapsulated ISUP involves the
       usage of proprietary parameters.

Appendix B. Acknowledgments

   We thank Andrew Dugan, Rob Maidhof, Dave Martin, Adam Roach, Jonathan
   Rosenberg, Dean Willis, Robert F.  Penfield, Steve Donovan, Allison
   Mankin, Scott Bradner and Steve Bellovin for their valuable comments.

   The original 'SIP+' proposal for interconnecting portions of the PSTN
   with SIP bridging was developed by Eric Zimmerer.

Authors' Addresses

   Aparna Vemuri-Pattisam
   Qwest Communications
   6000 Parkwood Pl
   Dublin, OH  43016 US
   EMail: Aparna.Vemuri@Qwest.com

   Jon Peterson
   NeuStar, Inc.
   1800 Sutter St
   Suite 570
   Concord, CA  94520 US
   Phone: +1 925/363-8720
   EMail: jon.peterson@neustar.biz
   URI:   http://www.neustar.biz/

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