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rfc:rfc3372

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

 Copyright (C) The Internet Society (2002).  All Rights Reserved.

Abstract

 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

 1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
 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
 4.1 Originator . . . . . . . . . . . . . . . . . . . . . . . . . . 12
 4.2 Terminator . . . . . . . . . . . . . . . . . . . . . . . . . . 13
 4.3 Intermediary . . . . . . . . . . . . . . . . . . . . . . . . . 14
 4.4 Behavioral Requirements Summary  . . . . . . . . . . . . . . . 15
 5.  Components of the SIP-T Protocol . . . . . . . . . . . . . . . 16
 5.1 Core SIP . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
 5.2 Encapsulation  . . . . . . . . . . . . . . . . . . . . . . . . 16
 5.3 Translation  . . . . . . . . . . . . . . . . . . . . . . . . . 16

Vemuri & Peterson Best Current Practice [Page 1] RFC 3372 SIP-T September 2002

 5.4 Support for mid-call signaling . . . . . . . . . . . . . . . . 17
 6.  SIP Content Negotiation  . . . . . . . . . . . . . . . . . . . 17
 7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
 8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 20
 9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
 10  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
 A.  Notes  . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
 B.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 21
 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
 Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 23

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

Vemuri & Peterson Best Current Practice [Page 2] RFC 3372 SIP-T September 2002

 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.

Vemuri & Peterson Best Current Practice [Page 3] RFC 3372 SIP-T September 2002

 Problem definition: To provide ISUP transparency across SS7-SIP
 interworking
 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
          Requirements
 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.

Vemuri & Peterson Best Current Practice [Page 4] RFC 3372 SIP-T September 2002

 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

       gateways).
 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
    phone.

Vemuri & Peterson Best Current Practice [Page 5] RFC 3372 SIP-T September 2002

  • * * * * * * ——- * * |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), Vemuri & Peterson Best Current Practice [Page 6] RFC 3372 SIP-T September 2002 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 Vemuri & Peterson Best Current Practice [Page 7] RFC 3372 SIP-T September 2002 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. Vemuri & Peterson Best Current Practice [Page 8] RFC 3372 SIP-T September 2002 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——>| | |====================Conversation=================| |——-REL——>| | | | |←—–RLC——-|——BYE——>| | | | | |—–REL—–>| | |←—200 OK—–| | | | | |←—RLC——| | | | | | 3.2 PSTN origination - IP termination * * * * * * * * * * |—-| |—–| /|MGC | VoIP Network |proxy|\ / —- —– \ / * * \ / * * \ / * * \ ——– * * ————- | PSTN | | SIP phone | ——– * ————- Figure 3: PSTN origination - IP termination Vemuri & Peterson Best Current Practice [Page 9] RFC 3372 SIP-T September 2002 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——-| | |←——200 OK——-| | |←—ANM—–| | | | |———ACK——–>| | | | |———ACK——–>| |=====================Conversation========================| |—–REL—→| | | | |———-BYE——→| | |←—RLC—–| |———BYE——–>| | | |←——200 OK——-| | |←——200 OK——-| | | | | | Vemuri & Peterson Best Current Practice [Page 10] RFC 3372 SIP-T September 2002 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
 parameters.
 A simple call-flow illustrating the different legs in the call is as
 shown below.

Vemuri & Peterson Best Current Practice [Page 11] RFC 3372 SIP-T September 2002

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

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.

Vemuri & Peterson Best Current Practice [Page 12] RFC 3372 SIP-T September 2002

 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.

Vemuri & Peterson Best Current Practice [Page 13] RFC 3372 SIP-T September 2002

 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:

Vemuri & Peterson Best Current Practice [Page 14] RFC 3372 SIP-T September 2002

 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
 elements

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
 terminate.
 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.

Vemuri & Peterson Best Current Practice [Page 15] RFC 3372 SIP-T September 2002

 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
 requirements.

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].

Vemuri & Peterson Best Current Practice [Page 16] RFC 3372 SIP-T September 2002

 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.

Vemuri & Peterson Best Current Practice [Page 17] RFC 3372 SIP-T September 2002

 1.  Support for ISUP is optional.  Therefore, UA2 accepts the INVITE
     irrespective of whether it can process the ISUP.
 UA1                    UA2
 INVITE-->
    (Content-type:multipart/mixed;
    Content-type: application/sdp;
    Content-disposition: session; handling=required;
    Content-type: application/isup;
    Content-disposition: signal; handling=optional;)
                       <--18x
 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;)
                         <--415
                   (Accept: application/sdp)
 ACK-->
 INVITE-->
 (Content-type: application/sdp)
                         <--18x
 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.

Vemuri & Peterson Best Current Practice [Page 18] RFC 3372 SIP-T September 2002

 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;)
                      <--415
                (Accept: application/sdp)
 ACK-->
 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

Vemuri & Peterson Best Current Practice [Page 19] RFC 3372 SIP-T September 2002

 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

 [1]   Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
       Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
       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.
 [5]   Handley, M. and V. Jacobson, "SDP: Session Description
       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
       2000.
 [8]   Rosenberg, J., Salama, H. and M. Squire, "Telephony Routing
       over IP (TRIP)", RFC 3219, January 2002.
 [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.

Vemuri & Peterson Best Current Practice [Page 20] RFC 3372 SIP-T September 2002

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

Vemuri & Peterson Best Current Practice [Page 21] RFC 3372 SIP-T September 2002

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
        vaparna10@yahoo.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/

Vemuri & Peterson Best Current Practice [Page 22] RFC 3372 SIP-T September 2002

Full Copyright Statement

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 or assist in its implementation may be prepared, copied, published
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Acknowledgement

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Vemuri & Peterson Best Current Practice [Page 23]

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