<pre>Network Working Group                                          J. Elwell
Request for Comments: 4497                                       Siemens
BCP: 117                                                        F. Derks
Category: Best Current Practice                              NEC Philips
                                                               P. Mourot
                                                             O. Rousseau
                                                                 Alcatel
                                                                May 2006


  <span class="h1">Interworking between the Session Initiation Protocol (SIP) and QSIG</span>

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 (2006).

Abstract

   This document specifies interworking between the Session Initiation
   Protocol (SIP) and QSIG within corporate telecommunication networks
   (also known as enterprise networks).  SIP is an Internet
   application-layer control (signalling) protocol for creating,
   modifying, and terminating sessions with one or more participants.
   These sessions include, in particular, telephone calls.  QSIG is a
   signalling protocol for creating, modifying, and terminating
   circuit-switched calls (in particular, telephone calls) within
   Private Integrated Services Networks (PISNs).  QSIG is specified in a
   number of Ecma Standards and published also as ISO/IEC standards.

















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Table of Contents

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-4">4</a>
   <a href="#section-2">2</a>. Terminology .....................................................<a href="#page-5">5</a>
   <a href="#section-3">3</a>. Definitions .....................................................<a href="#page-5">5</a>
      <a href="#section-3.1">3.1</a>. External Definitions .......................................<a href="#page-5">5</a>
      <a href="#section-3.2">3.2</a>. Other definitions ..........................................<a href="#page-5">5</a>
           <a href="#section-3.2.1">3.2.1</a>. Corporate Telecommunication Network (CN) ............<a href="#page-5">5</a>
           <a href="#section-3.2.2">3.2.2</a>. Gateway .............................................<a href="#page-6">6</a>
           <a href="#section-3.2.3">3.2.3</a>. IP Network ..........................................<a href="#page-6">6</a>
           <a href="#section-3.2.4">3.2.4</a>. Media Stream ........................................<a href="#page-6">6</a>
           <a href="#section-3.2.5">3.2.5</a>. Private Integrated Services Network (PISN) ..........<a href="#page-6">6</a>
           3.2.6. Private Integrated Services Network Exchange
                  (PINX) ..............................................<a href="#page-6">6</a>
   <a href="#section-4">4</a>. Acronyms ........................................................<a href="#page-6">6</a>
   <a href="#section-5">5</a>. Background and Architecture .....................................<a href="#page-7">7</a>
   <a href="#section-6">6</a>. Overview .......................................................<a href="#page-10">10</a>
   <a href="#section-7">7</a>. General Requirements ...........................................<a href="#page-11">11</a>
   <a href="#section-8">8</a>. Message Mapping Requirements ...................................<a href="#page-12">12</a>
      <a href="#section-8.1">8.1</a>. Message Validation and Handling of Protocol Errors ........<a href="#page-12">12</a>
      <a href="#section-8.2">8.2</a>. Call Establishment from QSIG to SIP .......................<a href="#page-14">14</a>
           8.2.1. Call Establishment from QSIG to SIP Using
                  En Bloc Procedures .................................<a href="#page-14">14</a>
           8.2.2. Call Establishment from QSIG to SIP Using
                  Overlap Procedures .................................<a href="#page-16">16</a>
      <a href="#section-8.3">8.3</a>. Call Establishment from SIP to QSIG .......................<a href="#page-20">20</a>
           <a href="#section-8.3.1">8.3.1</a>. Receipt of SIP INVITE Request for a New Call .......<a href="#page-20">20</a>
           <a href="#section-8.3.2">8.3.2</a>. Receipt of QSIG CALL PROCEEDING Message ............<a href="#page-21">21</a>
           <a href="#section-8.3.3">8.3.3</a>. Receipt of QSIG PROGRESS Message ...................<a href="#page-22">22</a>
           <a href="#section-8.3.4">8.3.4</a>. Receipt of QSIG ALERTING Message ...................<a href="#page-22">22</a>
           8.3.5. Inclusion of SDP Information in a SIP 18x
                  Provisional Response ...............................<a href="#page-23">23</a>
           <a href="#section-8.3.6">8.3.6</a>. Receipt of QSIG CONNECT Message ....................<a href="#page-24">24</a>
           <a href="#section-8.3.7">8.3.7</a>. Receipt of SIP PRACK Request .......................<a href="#page-25">25</a>
           <a href="#section-8.3.8">8.3.8</a>. Receipt of SIP ACK Request .........................<a href="#page-25">25</a>
           8.3.9. Receipt of a SIP INVITE Request for a Call
                  Already Being ......................................<a href="#page-25">25</a>
      <a href="#section-8.4">8.4</a>. Call Clearing and Call Failure ............................<a href="#page-26">26</a>
           8.4.1. Receipt of a QSIG DISCONNECT, RELEASE, or
                  RELEASE COMPLETE ...................................<a href="#page-26">26</a>
           <a href="#section-8.4.2">8.4.2</a>. Receipt of a SIP BYE Request .......................<a href="#page-29">29</a>
           <a href="#section-8.4.3">8.4.3</a>. Receipt of a SIP CANCEL Request ....................<a href="#page-29">29</a>
           8.4.4. Receipt of a SIP 4xx-6xx Response to an
                  INVITE Request .....................................<a href="#page-29">29</a>
           <a href="#section-8.4.5">8.4.5</a>. Gateway-Initiated Call Clearing ....................<a href="#page-32">32</a>
      <a href="#section-8.5">8.5</a>. Request to Change Media Characteristics ...................<a href="#page-32">32</a>





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   <a href="#section-9">9</a>. Number Mapping .................................................<a href="#page-32">32</a>
      <a href="#section-9.1">9.1</a>. Mapping from QSIG to SIP ..................................<a href="#page-33">33</a>
           9.1.1. Using Information from the QSIG Called
                  Party Number Information Element ...................<a href="#page-33">33</a>
           9.1.2. Using Information from the QSIG Calling
                  Party Number Information Element ...................<a href="#page-33">33</a>
           9.1.3. Using Information from the QSIG Connected
                  Number Information Element .........................<a href="#page-35">35</a>
      <a href="#section-9.2">9.2</a>. Mapping from SIP to QSIG ..................................<a href="#page-36">36</a>
           9.2.1. Generating the QSIG Called Party Number
                  Information Element ................................<a href="#page-36">36</a>
           9.2.2. Generating the QSIG Calling Party Number
                  Information Element ................................<a href="#page-37">37</a>
           9.2.3. Generating the QSIG Connected Number
                  Information Element ................................<a href="#page-38">38</a>
   <a href="#section-10">10</a>. Requirements for Support of Basic Services ....................<a href="#page-39">39</a>
      10.1. Derivation of QSIG Bearer Capability Information
            Element ..................................................<a href="#page-39">39</a>
      <a href="#section-10.2">10.2</a>. Derivation of Media Type in SDP ..........................<a href="#page-39">39</a>
   <a href="#section-11">11</a>. Security Considerations .......................................<a href="#page-40">40</a>
      <a href="#section-11.1">11.1</a>. General ..................................................<a href="#page-40">40</a>
      <a href="#section-11.2">11.2</a>. Calls from QSIG to Invalid or Restricted Numbers .........<a href="#page-40">40</a>
      <a href="#section-11.3">11.3</a>. Abuse of SIP Response Code ...............................<a href="#page-41">41</a>
      <a href="#section-11.4">11.4</a>. Use of the To Header URI .................................<a href="#page-41">41</a>
      <a href="#section-11.5">11.5</a>. Use of the From Header URI ...............................<a href="#page-41">41</a>
      <a href="#section-11.6">11.6</a>. Abuse of Early Media .....................................<a href="#page-42">42</a>
      <a href="#section-11.7">11.7</a>. Protection from Denial-of-Service Attacks ................<a href="#page-42">42</a>
   <a href="#section-12">12</a>. Acknowledgements ..............................................<a href="#page-43">43</a>
   <a href="#section-13">13</a>. Normative References ..........................................<a href="#page-43">43</a>
   <a href="#appendix-A">Appendix A</a>. Example Message Sequences .............................<a href="#page-45">45</a>





















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<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   This document specifies signalling interworking between QSIG and the
   Session Initiation Protocol (SIP) in support of basic services within
   a corporate telecommunication network (CN) (also known as enterprise
   network).

   QSIG is a signalling protocol that operates between Private
   Integrated Services eXchanges (PINX) within a Private Integrated
   Services Network (PISN).  A PISN provides circuit-switched basic
   services and supplementary services to its users.  QSIG is specified
   in Ecma Standards; in particular, [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] (call control in support of
   basic services), [<a href="#ref-3" title="&quot;Private Integrated Services Network - Generic Functional Protocol for the Support of Supplementary Services - Inter-Exchange Signalling Procedures and Protocol&quot;">3</a>] (generic functional protocol for the support of
   supplementary services), and a number of standards specifying
   individual supplementary services.

   NOTE: The name QSIG was derived from the fact that it is used for
   signalling at the Q reference point.  The Q reference point is a
   point of demarcation between two PINXs.

   SIP is an application-layer protocol for establishing, terminating,
   and modifying multimedia sessions.  It is typically carried over IP
   [<a href="#ref-15" title="&quot;Internet Protocol&quot;">15</a>], [<a href="#ref-16" title="&quot;Internet Protocol, Version 6 (IPv6) Specification&quot;">16</a>].  Telephone calls are considered a type of multimedia
   session where just audio is exchanged.  SIP is defined in [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>].

   As the support of telephony within corporate networks evolves from
   circuit-switched technology to Internet technology, the two
   technologies will coexist in many networks for a period, perhaps
   several years.  Therefore, there is a need to be able to establish,
   modify, and terminate sessions involving a participant in the SIP
   network and a participant in the QSIG network.  Such calls are
   supported by gateways that perform interworking between SIP and QSIG.

   This document specifies SIP-QSIG signalling interworking for basic
   services that provide a bi-directional transfer capability for
   speech, DTMF, facsimile, and modem media between a PISN employing
   QSIG and a corporate IP network employing SIP.  Other aspects of
   interworking, e.g., the use of RTP and SDP, will differ according to
   the type of media concerned and are outside the scope of this
   specification.

   Call-related and call-independent signalling in support of
   supplementary services is outside the scope of this specification,
   but support for certain supplementary services (e.g., call transfer,
   call diversion) could be the subject of future work.






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   Interworking between QSIG and SIP permits a call originating at a
   user of a PISN to terminate at a user of a corporate IP network, or a
   call originating at a user of a corporate IP network to terminate at
   a user of a PISN.

   Interworking between a PISN employing QSIG and a public IP network
   employing SIP is outside the scope of this specification.  However,
   the functionality specified in this specification is in principle
   applicable to such a scenario when deployed in conjunction with other
   relevant functionality (e.g., number translation, security functions,
   etc.).

   This specification is applicable to any interworking unit that can
   act as a gateway between a PISN employing QSIG and a corporate IP
   network employing SIP.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Terminology</span>

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in <a href="./rfc2119">RFC 2119</a> [<a href="#ref-4" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">4</a>] and
   indicate requirement levels for compliant SIP implementations.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Definitions</span>

   For the purposes of this specification, the following definitions
   apply.

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  External Definitions</span>

   The definitions in [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] apply as appropriate.

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  Other definitions</span>

<span class="h4"><a class="selflink" id="section-3.2.1" href="#section-3.2.1">3.2.1</a>.  Corporate Telecommunication Network (CN)</span>

   Sets of privately-owned or carrier-provided equipment that are
   located at geographically dispersed locations and are interconnected
   to provide telecommunication services to a defined group of users.

   NOTE: A CN can comprise a PISN, a private IP network (intranet), or a
   combination of the two.









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<span class="h4"><a class="selflink" id="section-3.2.2" href="#section-3.2.2">3.2.2</a>.  Gateway</span>

   An entity that performs interworking between a PISN using QSIG and an
   IP network using SIP.

<span class="h4"><a class="selflink" id="section-3.2.3" href="#section-3.2.3">3.2.3</a>.  IP Network</span>

   A network (unless otherwise stated, a corporate network) offering
   connectionless packet-mode services based on the Internet Protocol
   (IP) as the network-layer protocol.

<span class="h4"><a class="selflink" id="section-3.2.4" href="#section-3.2.4">3.2.4</a>.  Media Stream</span>

   Audio or other user information transmitted in UDP packets, typically
   containing RTP, in a single direction between the gateway and a peer
   entity participating in a session established using SIP.

   NOTE: Normally a SIP session establishes a pair of media streams, one
   in each direction.

<span class="h4"><a class="selflink" id="section-3.2.5" href="#section-3.2.5">3.2.5</a>.  Private Integrated Services Network (PISN)</span>

   A CN or part of a CN that employs circuit-switched technology.

<span class="h4"><a class="selflink" id="section-3.2.6" href="#section-3.2.6">3.2.6</a>.  Private Integrated Services Network Exchange (PINX)</span>

   A PISN nodal entity comprising switching and call handling functions
   and supporting QSIG signalling in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>].

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Acronyms</span>

   DNS   Domain Name Service
   IP    Internet Protocol
   PINX  Private Integrated services Network eXchange
   PISN  Private Integrated Services Network
   RTP   Real-time Transport Protocol
   SCTP  Stream Control Transmission Protocol
   SDP   Session Description Protocol
   SIP   Session Initiation Protocol
   TCP   Transmission Control Protocol
   TLS   Transport Layer Security
   TU    Transaction User
   UA    User Agent
   UAC   User Agent Client
   UAS   User Agent Server
   UDP   User Datagram Protocol





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<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Background and Architecture</span>

   During the 1980s, corporate voice telecommunications adopted
   technology similar in principle to Integrated Services Digital
   Networks (ISDN).  Digital circuit switches, commonly known as Private
   Branch eXchanges (PBX) or more formally as Private Integrated
   services Network eXchanges (PINX) have been interconnected by digital
   transmission systems to form Private Integrated Services Networks
   (PISN).  These digital transmission systems carry voice or other
   payload in fixed-rate channels, typically 64 Kbit/s, and signalling
   in a separate channel.  A technique known as common channel
   signalling is employed, whereby a single signalling channel
   potentially controls a number of payload channels or bearer channels.
   A typical arrangement is a point-to-point transmission facility at T1
   or E1 rate providing a 64 Kbit/s signalling channel and 23 or 30
   bearer channels, respectively.  Other arrangements are possible and
   have been deployed, including the use of multiple transmission
   facilities for a signalling channel and its logically associated
   bearer channels.  Also, arrangements involving bearer channels at
   sub-64 Kbit/s have been deployed, where voice payload requires the
   use of codecs that perform compression.

   QSIG is the internationally-standardized message-based signalling
   protocol for use in networks as described above.  It runs in a
   signalling channel between two PINXs and controls calls on a number
   of logically associated bearer channels between the same two PINXs.
   The signalling channel and its logically associated bearer channels
   are collectively known as an inter-PINX link.  QSIG is independent of
   the type of transmission capabilities over which the signalling
   channel and bearer channels are provided.  QSIG is also independent
   of the transport protocol used to transport QSIG messages reliably
   over the signalling channel.

   QSIG provides a means for establishing and clearing calls that
   originate and terminate on different PINXs.  A call can be routed
   over a single inter-PINX link connecting the originating and
   terminating PINX, or over several inter-PINX links in series with
   switching at intermediate PINXs known as transit PINXs.  A call can
   originate or terminate in another network, in which case it enters or
   leaves the PISN environment through a gateway PINX.  Parties are
   identified by numbers, in accordance with either [<a href="#ref-17" title="&quot;The International Public Telecommunication Numbering Plan&quot;">17</a>] or a private
   numbering plan.  This basic call capability is specified in [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>].  In
   addition to basic call capability, QSIG specifies a number of further
   capabilities supporting the use of supplementary services in PISNs.

   More recently, corporate telecommunications networks have started to
   exploit IP in various ways.  One way is to migrate part of the
   network to IP using SIP.  This might, for example, be a new branch



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   office with a SIP proxy and SIP endpoints instead of a PINX.
   Alternatively, SIP equipment might be used to replace an existing
   PINX or PINXs.  The new SIP environment needs to interwork with the
   QSIG-based PISN in order to support calls originating in one
   environment and terminating in the other.  Interworking is achieved
   through a gateway.

   Interworking between QSIG and SIP at gateways can also be used where
   a SIP network interconnects different parts of a PISN, thereby
   allowing calls between the different parts.  A call can enter the SIP
   network at one gateway and leave at another.  Each gateway would
   behave in accordance with this specification.

   Another way of connecting two parts of a PISN would be to encapsulate
   QSIG signalling in SIP messages for calls between the two parts.
   This is outside the scope of this specification but could be the
   subject of future work.

   This document specifies signalling protocol interworking aspects of a
   gateway between a PISN employing QSIG signalling and an IP network
   employing SIP signalling.  The gateway appears as a PINX to other
   PINXs in the PISN.  The gateway appears as a SIP endpoint to other
   SIP entities in the IP network.  The environment is shown in Figure
   1.

        +------+   IP network                  PISN
        |      |
        |SIP   |                                             +------+
        |Proxy |                                            /|      |
        |      |                                           / |PINX  |
        +---+--+             *-----------+                /  |      |
            |                |           |        +-----+/   +------+
            |                |           |        |     |
            |                |           |        |PINX |
   ---+-----+-------+--------+  Gateway  +--------|     |
      |             |        |           |        |     |\
      |             |        |           |        +-----+ \
      |             |        |           |                 \ +------+
      |             |        |           |                  \|      |
   +--+---+      +--+---+    *-----------+                   |PINX  |
   |SIP   |      |SIP   |                                    |      |
   |End-  |      |End-  |                                    +------+
   |point |      |point |
   +------+      +------+

                          Figure 1: Environment





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   In addition to the signalling interworking functionality specified in
   this specification, it is assumed that the gateway also includes the
   following functionality:

   - one or more physical interfaces on the PISN side supporting one or
     more inter-PINX links, each link providing one or more constant bit
     rate channels for media streams and a reliable layer 2 connection
     (e.g., over a fixed rate physical channel) for transporting QSIG
     signalling messages; and

   - one or more physical interfaces on the IP network side supporting,
     through layer 1 and layer 2 protocols, IP as the network layer
     protocol and UDP [<a href="#ref-6" title="&quot;User Datagram Protocol&quot;">6</a>] and TCP [<a href="#ref-5" title="&quot;Transmission Control Protocol&quot;">5</a>] as transport layer protocols,
     these being used for the transport of SIP signalling messages and,
     in the case of UDP, also for media streams;

   - optionally the support of TLS [<a href="#ref-7" title="&quot;The TLS Protocol Version 1.0&quot;">7</a>] and/or SCTP [<a href="#ref-9" title="&quot;Stream Control Transmission Protocol&quot;">9</a>] as additional
     transport layer protocols on the IP network side, these being used
     for the transport of SIP signalling messages; and

   - a means of transferring media streams in each direction between the
     PISN and the IP network, including as a minimum packetization of
     media streams sent to the IP network and de-packetization of media
     streams received from the IP network.

   NOTE: [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] mandates support for both UDP and TCP for the transport of
   SIP messages and allows optional support for TLS and/or SCTP for this
   same purpose.

   The protocol model relevant to signalling interworking functionality
   of a gateway is shown in Figure 2.




















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   +---------------------------------------------------------+
   |                   Interworking function                 |
   |                                                         |
   +-----------------------+---------+-----------------------+
   |                       |         |                       |
   |        SIP            |         |                       |
   |                       |         |                       |
   +-----------------------+         |                       |
   |                       |         |                       |
   |  UDP/TCP/TLS/SCTP     |         |        QSIG           |
   |                       |         |                       |
   +-----------------------+         |                       |
   |                       |         |                       |
   |        IP             |         |                       |
   |                       |         |                       |
   +-----------------------+         +-----------------------+
   |    IP network         |         |        PISN           |
   |    lower layers       |         |    lower layers       |
   |                       |         |                       |
   +-----------------------+         +-----------------------+

                    Figure 2: Protocol model

   In Figure 2, the SIP box represents SIP syntax and encoding, the SIP
   transport layer, and the SIP transaction layer.  The Interworking
   function includes SIP Transaction User (TU) functionality.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Overview</span>

   The gateway maps received QSIG messages, where appropriate, to SIP
   messages and vice versa and maintains an association between a QSIG
   call and a SIP dialog.

   A call from QSIG to SIP is initiated when a QSIG SETUP message
   arrives at the gateway.  The QSIG SETUP message initiates QSIG call
   establishment, and an initial response message (e.g., CALL
   PROCEEDING) completes negotiation of the bearer channel to be used
   for that call.  The gateway then sends a SIP INVITE request, having
   translated the QSIG called party number to a URI suitable for
   inclusion in the Request-URI.  The SIP INVITE request and the
   resulting SIP dialog, if successfully established, are associated
   with the QSIG call.  The SIP 2xx response to the INVITE request is
   mapped to a QSIG CONNECT message, signifying answer of the call.
   During establishment, media streams established by SIP and SDP are
   connected to the bearer channel.






<span class="grey">Elwell, et al.           Best Current Practice                 [Page 10]</span></pre>
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<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   A call from SIP to QSIG is initiated when a SIP INVITE request
   arrives at the gateway.  The gateway sends a QSIG SETUP message to
   initiate QSIG call establishment, having translated the SIP Request-
   URI to a number suitable for use as the QSIG called party number.
   The resulting QSIG call is associated with the SIP INVITE request and
   with the eventual SIP dialog.  Receipt of an initial QSIG response
   message completes negotiation of the bearer channel to be used,
   allowing media streams established by SIP and SDP to be connected to
   that bearer channel.  The QSIG CONNECT message is mapped to a SIP 200
   OK response to the INVITE request.

   <a href="#appendix-A">Appendix A</a> gives examples of typical message sequences that can
   arise.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  General Requirements</span>

   In order to conform to this specification, a gateway SHALL support
   QSIG in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] as a gateway and SHALL support SIP in
   accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] as a UA.  In particular, the gateway SHALL
   support SIP syntax and encoding, the SIP transport layer, and the SIP
   transaction layer in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>].  In addition, the gateway
   SHALL support SIP TU behaviour for a UA in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>]
   except where stated otherwise in Sections <a href="#section-8">8</a>, <a href="#section-9">9</a>, and <a href="#section-10">10</a> of this
   specification.

   NOTE: [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] mandates that a SIP entity support both UDP and TCP as
   transport layer protocols for SIP messages.  Other transport layer
   protocols can also be supported.

   The gateway SHALL also support SIP reliable provisional responses in
   accordance with [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>] as a UA.

   NOTE: [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>] makes provision for recovering from loss of provisional
   responses (other than 100) to INVITE requests when using unreliable
   transport services in the IP network.  This is important for ensuring
   delivery of responses that map to essential QSIG messages.

   The gateway SHALL support SDP in accordance with [<a href="#ref-8" title="&quot;SDP: Session Description Protocol&quot;">8</a>] and its use in
   accordance with the offer/answer model in [<a href="#ref-12" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">12</a>].

   <a href="#section-9">Section 9</a> also specifies optional use of the Privacy header in
   accordance with [<a href="#ref-13" title="&quot;A Privacy Mechanism for the Session Initiation Protocol (SIP)&quot;">13</a>] and the P-Asserted-Identity header in accordance
   with [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>].

   The gateway SHALL support calls from QSIG to SIP and calls from SIP
   to QSIG.





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   SIP methods not defined in [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] or [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>] are outside the scope of this
   specification but could be the subject of other specifications for
   interworking with QSIG, e.g., for interworking in support of
   supplementary services.

   As a result of DNS lookup by the gateway in order to determine where
   to send a SIP INVITE request, a number of candidate destinations can
   be attempted in sequence.  The way in which this is handled by the
   gateway is outside the scope of this specification.  However, any
   behaviour specified in this document on receipt of a SIP 4xx or 5xx
   final response to an INVITE request SHOULD apply only when there are
   no more candidate destinations to try or when overlap signalling
   applies in the SIP network (see 8.2.2.2).

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Message Mapping Requirements</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  Message Validation and Handling of Protocol Errors</span>

   The gateway SHALL validate received QSIG messages in accordance with
   the requirements of [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and SHALL act in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] on
   detection of a QSIG protocol error.  The requirements of this section
   for acting on a received QSIG message apply only to a received QSIG
   message that has been successfully validated and that satisfies one
   of the following conditions:

   -the QSIG message is a SETUP message and indicates a destination in
   the IP network and a bearer capability for which the gateway is able
   to provide interworking; or

   -the QSIG message is a message other than SETUP and contains a call
   reference that identifies an existing call for which the gateway is
   providing interworking between QSIG and SIP.

   The processing of any valid QSIG message that does not satisfy any of
   these conditions is outside the scope of this specification.  Also,
   the processing of any QSIG message relating to call-independent
   signalling connections or connectionless transport, as specified in
   [<a href="#ref-3" title="&quot;Private Integrated Services Network - Generic Functional Protocol for the Support of Supplementary Services - Inter-Exchange Signalling Procedures and Protocol&quot;">3</a>], is outside the scope of this specification.

   If segmented QSIG messages are received, the gateway SHALL await
   receipt of all segments of a message and SHALL validate and act on
   the complete reassembled message.

   The gateway SHALL validate received SIP messages (requests and
   responses) in accordance with the requirements of [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] and SHALL act
   in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] on detection of a SIP protocol error.





<span class="grey">Elwell, et al.           Best Current Practice                 [Page 12]</span></pre>
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<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   Requirements of this section for acting on a received SIP message
   apply only to a received message that has been successfully validated
   and that satisfies one of the following conditions:

   - the SIP message is an INVITE request that contains no tag parameter
     in the To header field, does not match an ongoing transaction
     (i.e., is not a merged request; see Section 8.2.2.2 of [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>]), and
     indicates a destination in the PISN for which the gateway is able
     to provide interworking; or

   - the SIP message is a request that relates to an existing dialog
     representing a call for which the gateway is providing interworking
     between QSIG and SIP; or

   - the SIP message is a CANCEL request that relates to a received
     INVITE request for which the gateway is providing interworking with
     QSIG but for which the only response sent is informational (1xx),
     no dialog having been confirmed; or

   - the SIP message is a response to a request sent by the gateway in
     accordance with this section.

   The processing of any valid SIP message that does not satisfy any of
   these conditions is outside the scope of this specification.

   NOTE: These rules mean that an error detected in a received message
   will not be propagated to the other side of the gateway.  However,
   there can be an indirect impact on the other side of the gateway,
   e.g., the initiation of call clearing procedures.

   The gateway SHALL run QSIG protocol timers as specified in [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and
   SHALL act in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] if a QSIG protocol timer expires.
   Any other action on expiry of a QSIG protocol timer is outside the
   scope of this specification, except that if it results in the
   clearing of the QSIG call, the gateway SHALL also clear the SIP call
   in accordance with <a href="#section-8.4.5">Section 8.4.5</a>.

   The gateway SHALL run SIP protocol timers as specified in [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] and
   SHALL act in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] if a SIP protocol timer expires.
   Any other action on expiry of a SIP protocol timer is outside the
   scope of this specification, except that if it results in the
   clearing of the SIP call, the gateway SHALL also clear the QSIG call
   in accordance with <a href="#section-8.4.5">Section 8.4.5</a>.








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<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  Call Establishment from QSIG to SIP</span>

<span class="h4"><a class="selflink" id="section-8.2.1" href="#section-8.2.1">8.2.1</a>.  Call Establishment from QSIG to SIP Using En Bloc Procedures</span>

   The following procedures apply when the gateway receives a QSIG SETUP
   message containing a Sending Complete information element or the
   gateway receives a QSIG SETUP message and is able to determine that
   the number in the Called party number information element is
   complete.

   NOTE: In the absence of a Sending Complete information element, the
   means by which the gateway determines the number to be complete is an
   implementation matter.  It can involve knowledge of the numbering
   plan and/or use of inter-digit timer expiry.

<span class="h5"><a class="selflink" id="section-8.2.1.1" href="#section-8.2.1.1">8.2.1.1</a>.  Receipt of QSIG SETUP Message</span>

   On receipt of a QSIG SETUP message containing a number that the
   gateway determines to be complete in the Called party number
   information element, or containing a Sending complete information
   element and a number that could potentially be complete, the gateway
   SHALL map the QSIG SETUP message to a SIP INVITE request.  The
   gateway SHALL also send a QSIG CALL PROCEEDING message.

   The gateway SHALL generate the SIP Request-URI, To, and From fields
   in the SIP INVITE request in accordance with <a href="#section-9">Section 9</a>.  The gateway
   SHALL include in the INVITE request a Supported header containing
   option tag 100rel, to indicate support for [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>].

   The gateway SHALL include SDP offer information in the SIP INVITE
   request as described in <a href="#section-10">Section 10</a>.  It SHOULD also connect the
   incoming media stream to the user information channel of the inter-
   PINX link, to allow the caller to hear in-band tones or announcements
   and prevent speech clipping on answer.  Because of forking, the
   gateway may receive more than one media stream, in which case it
   SHOULD select one (e.g., the first received).  If the gateway is able
   to correlate an unselected media stream with a particular early
   dialog established using a reliable provisional response, it MAY use
   the UPDATE method [<a href="#ref-19" title="&quot;The Session Initiation Protocol (SIP) UPDATE Method&quot;">19</a>] to stop that stream and then use the UPDATE
   method to start that stream again if a 2xx response is received on
   that dialog.

   On receipt of a QSIG SETUP message containing a Sending complete
   information element and a number that the gateway determines to be
   incomplete in the Called party number information element, the
   gateway SHALL initiate QSIG call clearing procedures using cause
   value 28, "invalid number format (address incomplete)".




<span class="grey">Elwell, et al.           Best Current Practice                 [Page 14]</span></pre>
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   If information in the QSIG SETUP message is unsuitable for generating
   any of the mandatory fields in a SIP INVITE request (e.g., if a
   Request-URI cannot be derived from the QSIG Called party number
   information element) or for generating SDP information, the gateway
   SHALL NOT issue a SIP INVITE request and SHALL initiate QSIG call
   clearing procedures in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>].

<span class="h5"><a class="selflink" id="section-8.2.1.2" href="#section-8.2.1.2">8.2.1.2</a>.  Receipt of SIP 100 (Trying) Response to an INVITE Request</span>

   A SIP 100 response SHALL NOT trigger any QSIG messages.  It only
   serves the purpose of suppressing INVITE request retransmissions.

<span class="h5"><a class="selflink" id="section-8.2.1.3" href="#section-8.2.1.3">8.2.1.3</a>.  Receipt of SIP 18x provisional response to an INVITE request</span>

   The gateway SHALL map a received SIP 18x response to an INVITE
   request to a QSIG PROGRESS or ALERTING message based on the following
   conditions.

   - If a SIP 180 response is received and no QSIG ALERTING message has
   been sent, the gateway SHALL generate a QSIG ALERTING message.  The
   gateway MAY supply ring-back tone on the user information channel of
   the inter-PINX link, in which case the gateway SHALL include progress
   description number 8 in the QSIG ALERTING message.  Otherwise the
   gateway SHALL NOT include progress description number 8 in the QSIG
   ALERTING message unless the gateway is aware that in-band information
   (e.g., ring-back tone) is being transmitted.

   - If a SIP 181/182/183 response is received, no QSIG ALERTING message
   has been sent, and no message containing progress description number
   1 has been sent, the gateway SHALL generate a QSIG PROGRESS message
   containing progress description number 1.

   NOTE: This will ensure that QSIG timer T310 is stopped if running at
   the Originating PINX.

   In all other scenarios, the gateway SHALL NOT map the SIP 18x
   response to a QSIG message.

   If the SIP 18x response contains a Require header with option tag
   100rel, the gateway SHALL send back a SIP PRACK request in accordance
   with [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>].

<span class="h5"><a class="selflink" id="section-8.2.1.4" href="#section-8.2.1.4">8.2.1.4</a>.  Receipt of SIP 2xx Response to an INVITE Request</span>

   If the gateway receives a SIP 2xx response as the first SIP 2xx
   response to a SIP INVITE request, the gateway SHALL map the SIP 2xx
   response to a QSIG CONNECT message.  The gateway SHALL also send a
   SIP ACK request to acknowledge the 2xx response.  The gateway SHALL



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   NOT include any SDP information in the SIP ACK request.  If the
   gateway receives further 2xx responses, it SHALL respond to each in
   accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], SHOULD issue a BYE request for each, and SHALL
   NOT generate any further QSIG messages.

   Media streams will normally have been established in the IP network
   in each direction.  If so, the gateway SHALL connect the media
   streams to the corresponding user-information channel on the inter-
   PINX link if it has not already done so and stop any local ring-back
   tone.

   If the SIP 2xx response is received in response to the SIP PRACK
   request, the gateway SHALL NOT map this message to any QSIG message.

   NOTE: A SIP 2xx response to the INVITE request can be received later
   on a different dialog as a result of a forking proxy.

<span class="h5"><a class="selflink" id="section-8.2.1.5" href="#section-8.2.1.5">8.2.1.5</a>.  Receipt of SIP 3xx Response to an INVITE Request</span>

   On receipt of a SIP 3xx response to an INVITE request, the gateway
   SHALL act in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>].

   NOTE: This will normally result in sending a new SIP INVITE request.

   Unless the gateway supports the QSIG Call Diversion Supplementary
   Service, no QSIG message SHALL be sent.  The definition of Call
   Diversion Supplementary Service for QSIG to SIP interworking is
   beyond the scope of this specification.

<span class="h4"><a class="selflink" id="section-8.2.2" href="#section-8.2.2">8.2.2</a>.  Call Establishment from QSIG to SIP Using Overlap Procedures</span>

   SIP uses en bloc signalling, and it is strongly RECOMMENDED to avoid
   using overlap signalling in a SIP network.  A SIP/QSIG gateway
   dealing with overlap signalling SHOULD perform a conversion from
   overlap to en bloc signalling method using one or more of the
   following mechanisms:

      - timers;

      - numbering plan information;

      - the presence of a Sending complete information element in a
        received QSIG INFORMATION message.

   If the gateway performs a conversion from overlap to en bloc
   signalling in the SIP network, then the procedures defined in <a href="#section-8.2.2.1">Section</a>
   <a href="#section-8.2.2.1">8.2.2.1</a> SHALL apply.




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   However, for some applications it might be impossible to avoid using
   overlap signalling in the SIP network.  In this case, the procedures
   defined in <a href="#section-8.2.2.2">Section 8.2.2.2</a> SHALL apply.

<span class="h5"><a class="selflink" id="section-8.2.2.1" href="#section-8.2.2.1">8.2.2.1</a>.  En Bloc Signalling in SIP Network</span>

<span class="h6"><a class="selflink" id="section-8.2.2.1.1" href="#section-8.2.2.1.1">8.2.2.1.1</a>.  Receipt of QSIG SETUP Message</span>

   On receipt of a QSIG SETUP message containing no Sending complete
   information element and a number in the Called party number
   information element that the gateway cannot determine to be complete,
   the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start
   QSIG timer T302, and await further number digits.

<span class="h6"><a class="selflink" id="section-8.2.2.1.2" href="#section-8.2.2.1.2">8.2.2.1.2</a>.  Receipt of QSIG INFORMATION Message</span>

   On receipt of each QSIG INFORMATION message containing no Sending
   complete information element and containing a number that the gateway
   cannot determine to be complete, QSIG timer T302 SHALL be restarted.
   When QSIG timer T302 expires or a QSIG INFORMATION message containing
   a Sending complete information element is received, the gateway SHALL
   send a SIP INVITE request as described in <a href="#section-8.2.1.1">Section 8.2.1.1</a>.  The
   Request-URI and To fields (see <a href="#section-9">Section 9</a>) SHALL be generated from the
   concatenation of information in the Called party number information
   element in the received QSIG SETUP and INFORMATION messages.  The
   gateway SHALL also send a QSIG CALL PROCEEDING message.

<span class="h6"><a class="selflink" id="section-8.2.2.1.3" href="#section-8.2.2.1.3">8.2.2.1.3</a>.  Receipt of SIP Responses to INVITE Requests</span>

   SIP responses to INVITE requests SHALL be mapped as described in
   8.2.1.

<span class="h5"><a class="selflink" id="section-8.2.2.2" href="#section-8.2.2.2">8.2.2.2</a>.  Overlap Signalling in SIP Network</span>

   The procedures below for using overlap signalling in the SIP network
   are in accordance with the principles described in [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>] for using
   overlap sending when interworking with ISDN User Part (ISUP).  In
   [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>], there is discussion of some potential problems arising from the
   use of overlap sending in the SIP network.  These potential problems
   are applicable also in the context of QSIG-SIP interworking and can
   be avoided if overlap sending in the QSIG network is terminated at
   the gateway, in accordance with <a href="#section-8.2.2.1">Section 8.2.2.1</a>.  The procedures
   below should be used only where it is not feasible to use the
   procedures of <a href="#section-8.2.2.1">Section 8.2.2.1</a>.







<span class="grey">Elwell, et al.           Best Current Practice                 [Page 17]</span></pre>
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<span class="h6"><a class="selflink" id="section-8.2.2.2.1" href="#section-8.2.2.2.1">8.2.2.2.1</a>.  Receipt of QSIG SETUP Message</span>

   On receipt of a QSIG SETUP message containing no Sending complete
   information element and a number in the Called party number
   information element that the gateway cannot determine to be complete,
   the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message and
   start QSIG timer T302.  If the QSIG SETUP message contains the
   minimum number of digits required to route the call in the IP
   network, the gateway SHALL send a SIP INVITE request as specified in
   <a href="#section-8.2.1.1">Section 8.2.1.1</a>.  Otherwise, the gateway SHALL wait for more digits
   to arrive in QSIG INFORMATION messages.

<span class="h6"><a class="selflink" id="section-8.2.2.2.2" href="#section-8.2.2.2.2">8.2.2.2.2</a>.  Receipt of QSIG INFORMATION Message</span>

   On receipt of a QSIG INFORMATION message, the gateway SHALL handle
   the QSIG timer T302 in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>].

   NOTE: [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] requires the QSIG timer to be stopped if the INFORMATION
   message contains a Sending complete information element or to be
   restarted otherwise.

   Further behaviour of the gateway SHALL depend on whether or not it
   has already sent a SIP INVITE request.  If the gateway has not sent a
   SIP INVITE request and it now has the minimum number of digits
   required to route the call, it SHALL send a SIP INVITE request as
   specified in <a href="#section-8.2.2.1.2">Section 8.2.2.1.2</a>.  If the gateway still does not have
   the minimum number of digits required, it SHALL wait for more QSIG
   INFORMATION messages to arrive.

   If the gateway has already sent one or more SIP INVITE requests,
   whether or not final responses to those requests have been received,
   it SHALL send a new SIP INVITE request in accordance with Section 3.2
   of [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>].  The updated Request-URI and To fields (see <a href="#section-9">Section 9</a>) SHALL
   be generated from the concatenation of information in the Called
   party number information element in the received QSIG SETUP and
   INFORMATION messages.

   NOTE: [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>] requires the new request to have the same Call-ID and the
   same From header (including tag) as in the previous INVITE request.
   [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>] recommends that the CSeq header should contain a value higher
   than that in the previous INVITE request.

<span class="h6"><a class="selflink" id="section-8.2.2.2.3" href="#section-8.2.2.2.3">8.2.2.2.3</a>.  Receipt of SIP 100 (Trying) Response to an INVITE Request</span>

   The requirements of <a href="#section-8.2.1.2">Section 8.2.1.2</a> SHALL apply.






<span class="grey">Elwell, et al.           Best Current Practice                 [Page 18]</span></pre>
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<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h6"><a class="selflink" id="section-8.2.2.2.4" href="#section-8.2.2.2.4">8.2.2.2.4</a>.  Receipt of SIP 18x Provisional Response to an INVITE Request</span>

   The requirements of <a href="#section-8.2.1.3">Section 8.2.1.3</a> SHALL apply.

<span class="h6"><a class="selflink" id="section-8.2.2.2.5" href="#section-8.2.2.2.5">8.2.2.2.5</a>.  Receipt of SIP 2xx Response to an INVITE Request</span>

   The requirements of <a href="#section-8.2.1.4">Section 8.2.1.4</a> SHALL apply.  In addition, the
   gateway SHALL send a SIP CANCEL request in accordance with <a href="#section-3.4">Section</a>
   <a href="#section-3.4">3.4</a> of [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>] to cancel any SIP INVITE transactions for which no final
   response has been received.

<span class="h6"><a class="selflink" id="section-8.2.2.2.6" href="#section-8.2.2.2.6">8.2.2.2.6</a>.  Receipt of SIP 3xx Response to an INVITE Request</span>

   The requirements of <a href="#section-8.2.1.5">Section 8.2.1.5</a> SHALL apply.

<span class="h6"><a class="selflink" id="section-8.2.2.2.7" href="#section-8.2.2.2.7">8.2.2.2.7</a>.  Receipt of a SIP 4xx, 5xx, or 6xx Final Response to an</span>
<span class="h6">            INVITE Request</span>

   On receipt of a SIP 4xx, 5xx, or 6xx final response to an INVITE
   request, the gateway SHALL send back a SIP ACK request.  Unless the
   gateway is able to retry the INVITE request to avoid the problem
   (e.g., by supplying authentication in the case of a 401 or 407
   response), the gateway SHALL also send a QSIG DISCONNECT message
   (8.4.4) if no further QSIG INFORMATION messages are expected and
   final responses have been received to all transmitted SIP INVITE
   requests.

   NOTE: Further QSIG INFORMATION messages will not be expected after
   QSIG timer T302 has expired or after a Sending complete information
   element has been received.

   In all other cases, the receipt of a SIP 4xx, 5xx, or 6xx final
   response to an INVITE request SHALL NOT trigger the sending of any
   QSIG message.

   NOTE: If further QSIG INFORMATION messages arrive, these will result
   in further SIP INVITE requests being sent, one of which might result
   in successful call establishment.  For example, initial INVITE
   requests might produce 484 (Address Incomplete) or 404 (Not Found)
   responses because the Request-URIs derived from incomplete numbers
   cannot be routed, yet a subsequent INVITE request with a routable
   Request-URI might produce a 2xx final response or a more meaningful
   4xx, 5xx, or 6xx final response.








<span class="grey">Elwell, et al.           Best Current Practice                 [Page 19]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-20" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h6"><a class="selflink" id="section-8.2.2.2.8" href="#section-8.2.2.2.8">8.2.2.2.8</a>.  Receipt of Multiple SIP Responses to an INVITE Request</span>

   Section 3.3 of [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>] applies.

<span class="h6"><a class="selflink" id="section-8.2.2.2.9" href="#section-8.2.2.2.9">8.2.2.2.9</a>.  Cancelling Pending SIP INVITE Transactions</span>

   As stated in Section 3.4 of [<a href="#ref-18" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">18</a>], when a gateway sends a new SIP
   INVITE request containing new digits, it SHOULD NOT send a SIP CANCEL
   request to cancel a previous SIP INVITE transaction that has not had
   a final response.  This SIP CANCEL request could arrive at an egress
   gateway before the new SIP INVITE request and trigger premature call
   clearing.

   NOTE: Previous SIP INVITE transactions can be expected to result in
   SIP 4xx class responses, which terminate the transaction.  In <a href="#section-8.2.2.2.5">Section</a>
   <a href="#section-8.2.2.2.5">8.2.2.2.5</a>, there is provision for cancelling any transactions still
   in progress after a SIP 2xx response has been received.

<span class="h6"><a class="selflink" id="section-8.2.2.2.10" href="#section-8.2.2.2.10">8.2.2.2.10</a>.  QSIG Timer T302 Expiry</span>

   If QSIG timer T302 expires and the gateway has received 4xx, 5xx, or
   6xx responses to all transmitted SIP INVITE requests, the gateway
   SHALL send a QSIG DISCONNECT message.  If T302 expires and the
   gateway has not received 4xx, 5xx, or 6xx responses to all
   transmitted SIP INVITE requests, the gateway SHALL ignore any further
   QSIG INFORMATION messages but SHALL NOT send a QSIG DISCONNECT
   message at this stage.

   NOTE: A QSIG DISCONNECT request will be sent when all outstanding SIP
   INVITE requests have received 4xx, 5xx, or 6xx responses.

<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a>.  Call Establishment from SIP to QSIG</span>

<span class="h4"><a class="selflink" id="section-8.3.1" href="#section-8.3.1">8.3.1</a>.  Receipt of SIP INVITE Request for a New Call</span>

   On receipt of a SIP INVITE request for a new call, if a suitable
   channel is available on the inter-PINX link, the gateway SHALL
   generate a QSIG SETUP message from the received SIP INVITE request.
   The gateway SHALL generate the Called party number and Calling party
   number information elements in accordance with <a href="#section-9">Section 9</a> and SHALL
   generate the Bearer capability information element in accordance with
   <a href="#section-10">Section 10</a>.  If the gateway can determine that the number placed in
   the Called party number information element is complete, the gateway
   MAY include the Sending complete information element.

   NOTE: The means by which the gateway determines the number to be
   complete is an implementation matter.  It can involve knowledge of
   the numbering plan and/or use of the inter-digit timer.



<span class="grey">Elwell, et al.           Best Current Practice                 [Page 20]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-21" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   The gateway SHOULD send a SIP 100 (Trying) response.

   If information in the SIP INVITE request is unsuitable for generating
   any of the mandatory information elements in a QSIG SETUP message
   (e.g., if a QSIG Called party number information element cannot be
   derived from SIP Request-URI field) or if no suitable channel is
   available on the inter-PINX link, the gateway SHALL NOT issue a QSIG
   SETUP message and SHALL send a SIP 4xx, 5xx, or 6xx response.  If no
   suitable channel is available, the gateway should use response code
   503 (Service Unavailable).

   If the SIP INVITE request does not contain SDP information and does
   not contain either a Required header or a Supported header with
   option tag 100rel, the gateway SHOULD still proceed as above,
   although an implementation can instead send a SIP 488 (Not Acceptable
   Here) response, in which case it SHALL NOT issue a QSIG SETUP
   message.

   NOTE: The absence of SDP offer information in the SIP INVITE request
   means that the gateway might need to send SDP offer information in a
   provisional response and receive SDP answer information in a SIP
   PRACK request (in accordance with [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>]) in order to ensure that tones
   and announcements from the PISN are transmitted. SDP offer
   information cannot be sent in an unreliable provisional response
   because SDP answer information would need to be returned in a SIP
   PRACK request.  The recommendation above still to proceed with call
   establishment in this situation reflects the desire to maximise the
   chances of a successful call.  However, if important in-band
   information is likely to be denied in this situation, a gateway can
   choose not to proceed.

   NOTE: If SDP offer information is present in the INVITE request, the
   issuing of a QSIG SETUP message is not dependent on the presence of a
   Required header or a Supported header with option tag 100rel.

   On receipt of a SIP INVITE request relating to a call that has
   already been established from SIP to QSIG, the procedures of 8.3.9
   SHALL apply.

<span class="h4"><a class="selflink" id="section-8.3.2" href="#section-8.3.2">8.3.2</a>.  Receipt of QSIG CALL PROCEEDING Message</span>

   The receipt of a QSIG CALL PROCEEDING message SHALL NOT result in any
   SIP message being sent.








<span class="grey">Elwell, et al.           Best Current Practice                 [Page 21]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-22" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="section-8.3.3" href="#section-8.3.3">8.3.3</a>.  Receipt of QSIG PROGRESS Message</span>

   A QSIG PROGRESS message can be received in the event of interworking
   on the remote side of the PISN or if the PISN is unable to complete
   the call and generates an in-band tone or announcement.  In the
   latter case, a Cause information element is included in the QSIG
   PROGRESS message.

   The gateway SHALL map a received QSIG PROGRESS message to a SIP 183
   (Session Progress) response to the INVITE request.  If the SIP INVITE
   request contained either a Require header or a Supported header with
   option tag 100rel, the gateway SHALL include in the SIP 183 response
   a Require header with option tag 100rel.

   NOTE: In accordance with [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>], inclusion of option tag 100rel in a
   provisional response instructs the UAC to acknowledge the provisional
   response by sending a PRACK request.  [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>] also specifies procedures
   for repeating a provisional response with option tag 100rel if no
   PRACK is received.

   If the QSIG PROGRESS message contained a Progress indicator
   information element with Progress description number 1 or 8, the
   gateway SHALL connect the media streams to the corresponding user
   information channel of the inter-PINX link if it has not already done
   so, provided that SDP answer information is included in the
   transmitted SIP response to the INVITE request or has already been
   sent or received.  Inclusion of SDP offer or answer information in
   the 183 provisional response SHALL be in accordance with <a href="#section-8.3.5">Section</a>
   <a href="#section-8.3.5">8.3.5</a>.

   If the QSIG PROGRESS message is received with a Cause information
   element, the gateway SHALL either wait until the tone/announcement is
   complete or has been applied for sufficient time before initiating
   call clearing, or wait for a SIP CANCEL request.  If call clearing is
   initiated, the cause value in the QSIG PROGRESS message SHALL be used
   to derive the response to the SIP INVITE request in accordance with
   Table 1.

<span class="h4"><a class="selflink" id="section-8.3.4" href="#section-8.3.4">8.3.4</a>.  Receipt of QSIG ALERTING Message</span>

   The gateway SHALL map a QSIG ALERTING message to a SIP 180 (Ringing)
   response to the INVITE request.  If the SIP INVITE request contained
   either a Require header or a Supported header with option tag 100rel,
   the gateway SHALL include in the SIP 180 response a Require header
   with option tag 100rel.






<span class="grey">Elwell, et al.           Best Current Practice                 [Page 22]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-23" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   NOTE: In accordance with [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>], inclusion of option tag 100rel in a
   provisional response instructs the UAC to acknowledge the provisional
   response by sending a PRACK request.  [<a href="#ref-11" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">11</a>] also specifies procedures
   for repeating a provisional response with option tag 100rel if no
   PRACK is received.

   If the QSIG ALERTING message contained a Progress indicator
   information element with Progress description number 1 or 8, the
   gateway SHALL connect the media streams to the corresponding user
   information channel of the inter-PINX link if it has not already done
   so, provided that SDP answer information is included in the
   transmitted SIP response or has already been sent or received.
   Inclusion of SDP offer or answer information in the 180 provisional
   response SHALL be in accordance with <a href="#section-8.3.5">Section 8.3.5</a>.

<span class="h4"><a class="selflink" id="section-8.3.5" href="#section-8.3.5">8.3.5</a>.  Inclusion of SDP Information in a SIP 18x Provisional Response</span>

   When sending a SIP 18x provisional response to the INVITE request, if
   a QSIG message containing a Progress indicator information element
   with progress description number 1 or 8 has been received the gateway
   SHALL include SDP information.  Otherwise, the gateway MAY include
   SDP information.  If SDP information is included, it shall be in
   accordance with the following rules.

   If the SIP INVITE request contained a Required or Supported header
   with option tag 100rel, and if SDP offer and answer information has
   already been exchanged, no SDP information SHALL be included in the
   SIP 18x provisional response.

   If the SIP INVITE request contained a Required or Supported header
   with option tag 100rel, and if SDP offer information was received in
   the SIP INVITE request but no SDP answer information has been sent,
   SDP answer information SHALL be included in the SIP 18x provisional
   response.

   If the SIP INVITE request contained a Required or Supported header
   with option tag 100rel, and if no SDP offer information was received
   in the SIP INVITE request and no SDP offer information has already
   been sent, SDP offer information SHALL be included in the SIP 18x
   provisional response.

   NOTE: In this case, SDP answer information can be expected in the SIP
   PRACK.

   If the SIP INVITE request contained neither a Required nor a
   Supported header with option tag 100rel, SDP answer information SHALL
   be included in the SIP 18x provisional response.




<span class="grey">Elwell, et al.           Best Current Practice                 [Page 23]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-24" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   NOTE: Because the provisional response is unreliable, SDP answer
   information needs to be repeated in each provisional response and in
   the final SIP 2xx response.

   NOTE: If the SIP INVITE request contained no SDP offer information
   and neither a Required nor a Supported header with option tag 100rel,
   it should have been rejected in accordance with <a href="#section-8.3.1">Section 8.3.1</a>.

<span class="h4"><a class="selflink" id="section-8.3.6" href="#section-8.3.6">8.3.6</a>.  Receipt of QSIG CONNECT Message</span>

   The gateway SHALL map a QSIG CONNECT message to a SIP 200 (OK) final
   response for the SIP INVITE request.  The gateway SHALL also send a
   QSIG CONNECT ACKNOWLEDGE message.

   If the SIP INVITE request contained a Required or Supported header
   with option tag 100rel, and if SDP offer and answer information has
   already been exchanged, no SDP information SHALL be included in the
   SIP 200 response.

   If the SIP INVITE request contained a Required or Supported header
   with option tag 100rel, and if SDP offer information was received in
   the SIP INVITE request but no SDP answer information has been sent,
   SDP answer information SHALL be included in the SIP 200 response.

   If the SIP INVITE request contained a Required or Supported header
   with option tag 100rel, and if no SDP offer information was received
   in the SIP INVITE request and no SDP offer information has already
   been sent, SDP offer information SHALL be included in the SIP 200
   response.

   NOTE: In this case, SDP answer information can be expected in the SIP
   ACK.

   If the SIP INVITE request contained neither a Required nor a
   Supported header with option tag 100rel, SDP answer information SHALL
   be included in the SIP 200 response.

   NOTE: Because the provisional response is unreliable, SDP answer
   information needs to be repeated in each provisional response and in
   the final 2xx response.

   NOTE: If the SIP INVITE request contained no SDP offer information
   and neither a Required nor a Supported header with option tag 100rel,
   it may have been rejected in accordance with <a href="#section-8.3.1">Section 8.3.1</a>.







<span class="grey">Elwell, et al.           Best Current Practice                 [Page 24]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-25" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   The gateway SHALL connect the media streams to the corresponding user
   information channel of the inter-PINX link if it has not already done
   so, provided that SDP answer information is included in the
   transmitted SIP response or has already been sent or received.

<span class="h4"><a class="selflink" id="section-8.3.7" href="#section-8.3.7">8.3.7</a>.  Receipt of SIP PRACK Request</span>

   The receipt of a SIP PRACK request acknowledging a reliable
   provisional response SHALL NOT result in any QSIG message being sent.
   The gateway SHALL send back a SIP 200 (OK) response to the SIP PRACK
   request.

   If the SIP PRACK contains SDP answer information and a QSIG message
   containing a Progress indicator information element with progress
   description number 1 or 8 has been received, the gateway SHALL
   connect the media streams to the corresponding user information
   channel of the inter-PINX link.

<span class="h4"><a class="selflink" id="section-8.3.8" href="#section-8.3.8">8.3.8</a>.  Receipt of SIP ACK Request</span>

   The receipt of a SIP ACK request SHALL NOT result in any QSIG message
   being sent.

   If the SIP ACK contains SDP answer information, the gateway SHALL
   connect the media streams to the corresponding user information
   channel of the inter-PINX link if it has not already done so.

<span class="h4"><a class="selflink" id="section-8.3.9" href="#section-8.3.9">8.3.9</a>.  Receipt of a SIP INVITE Request for a Call Already Being</span>
<span class="h4">        Established</span>

   A gateway can receive a call from SIP using overlap procedures.  This
   should occur when the UAC for the INVITE request is a gateway from a
   network that employs overlap procedures (e.g., an ISUP gateway or
   another QSIG gateway) and the gateway has not absorbed overlap.

   For a call from SIP using overlap procedures, the gateway will
   receive multiple SIP INVITE requests that belong to the same call but
   have different Request-URI and To fields.  Each SIP INVITE request
   belongs to a different dialog.

   A SIP INVITE request is considered to be for the purpose of overlap
   sending if, compared to a previously received SIP INVITE request, it
   has:

      - the same Call-ID header;
      - the same From header (including the tag);
      - no tag in the To header;




<span class="grey">Elwell, et al.           Best Current Practice                 [Page 25]</span></pre>
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<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


      - an updated Request-URI from which can be derived a called party
        number with a superset of the digits derived from the previously
        received SIP INVITE request;

      and if

      - the gateway has not yet sent a final response other than 484 to
        the previously received SIP INVITE request.

   If a gateway receives a SIP INVITE request for the purpose of overlap
   sending, it SHALL generate a QSIG INFORMATION message using the call
   reference of the existing QSIG call instead of a new QSIG SETUP
   message and containing only the additional digits in the Called party
   number information element.  It SHALL also respond to the SIP INVITE
   request received previously with a SIP 484 Address Incomplete
   response.

   If a gateway receives a SIP INVITE request that meets all of the
   conditions for a SIP INVITE request for the purpose of overlap
   sending except the condition concerning the Request-URI, the gateway
   SHALL respond to the new request with a SIP 485 (Ambiguous) response.

<span class="h3"><a class="selflink" id="section-8.4" href="#section-8.4">8.4</a>.  Call Clearing and Call Failure</span>

<span class="h4"><a class="selflink" id="section-8.4.1" href="#section-8.4.1">8.4.1</a>.  Receipt of a QSIG DISCONNECT, RELEASE, or RELEASE COMPLETE</span>
<span class="h4">        Message</span>

   On receipt of QSIG DISCONNECT, RELEASE, or RELEASE COMPLETE message
   as the first QSIG call clearing message, gateway behaviour SHALL
   depend on the state of call establishment.

   1) If the gateway has sent a SIP 200 (OK) response to a SIP INVITE
      request and received a SIP ACK request, or if it has received a
      SIP 200 (OK) response to a SIP INVITE request and sent a SIP ACK
      request, the gateway SHALL send a SIP BYE request to clear the
      call.

   2) If the gateway has sent a SIP 200 (OK) response to a SIP INVITE
      request (indicating that call establishment is complete) but has
      not received a SIP ACK request, the gateway SHALL wait until a SIP
      ACK is received and then send a SIP BYE request to clear the call.

   3) If the gateway has sent a SIP INVITE request and received a SIP
      provisional response but not a SIP final response, the gateway
      SHALL send a SIP CANCEL request to clear the call.






<span class="grey">Elwell, et al.           Best Current Practice                 [Page 26]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-27" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


      NOTE 1: In accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], if after sending a SIP CANCEL
      request a SIP 2xx response is received to the SIP INVITE request,
      the gateway will need to send a SIP BYE request.

   4) If the gateway has sent a SIP INVITE request but received no SIP
      response, the gateway SHALL NOT send a SIP message.  If a SIP
      final or provisional response is subsequently received, the
      gateway SHALL then act in accordance with 1, 2, or 3 above,
      respectively.

   5) If the gateway has received a SIP INVITE request but not sent a
      SIP final response, the gateway SHALL send a SIP final response
      chosen according to the cause value in the received QSIG message
      as specified in Table 1.  SIP response 500 (Server internal error)
      SHALL be used as the default for cause values not shown in
      Table 1.

   NOTE 2: It is not necessarily appropriate to map some QSIG cause
   values to SIP messages because these cause values are meaningful only
   at the gateway.  A good example of this is cause value 44, "Requested
   circuit or channel not available", which signifies that the channel
   number in the transmitted QSIG SETUP message was not acceptable to
   the peer PINX.  The appropriate behavior in this case is for the
   gateway to send another SETUP message indicating a different channel
   number.  If this is not possible, the gateway should treat it either
   as a congestion situation (no channels available; see <a href="#section-8.3.1">Section 8.3.1</a>)
   or as a gateway failure situation (in which case the default SIP
   response code applies).

   In all cases, the gateway SHALL also disconnect media streams, if
   established, and allow QSIG and SIP signalling to complete in
   accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], respectively.



















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 27]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-28" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   Table 1: Mapping of QSIG Cause Value to SIP 4xx-6xx responses to an
   INVITE request

   QSIG Cause value               SIP response
   ----------------------------------------------------------------
   1  Unallocated number          404 Not found
   2  No route to specified       404 Not found
      transit network
   3  No route to destination     404 Not found
   16 Normal call clearing        (NOTE 3)
   17 User busy                   486 Busy here
   18 No user responding          408 Request timeout
   19 No answer from the user     480 Temporarily unavailable
   20 Subscriber absent           480 Temporarily unavailable
   21 Call rejected               603 Decline, if location field
                                      in Cause information element
                                      indicates user.  Otherwise:
                                      403 Forbidden
   22 Number changed              301 Moved permanently, if
                                      information in diagnostic field
                                      of Cause information element is
                                      suitable for generating a SIP
                                      Contact header.  Otherwise:
                                      410 Gone
   23 Redirection to new          410 Gone
      destination
   27 Destination out of order    502 Bad gateway
   28 Address incomplete          484 Address incomplete
   29 Facility rejected           501 Not implemented
   31 Normal, unspecified         480 Temporarily unavailable
   34 No circuit/channel          503 Service unavailable
      available
   38 Network out of order        503 Service unavailable
   41 Temporary failure           503 Service unavailable
   42 Switching equipment         503 Service unavailable
      congestion
   47 Resource unavailable,       503 Service unavailable
      unspecified
   55 Incoming calls barred       403 Forbidden
      within CUG
   57 Bearer capability not       403 Forbidden
      authorized
   58 Bearer capability not       503 Service unavailable
      presently available
   65 Bearer capability not       488 Not acceptable here (NOTE 4)
      implemented
   69 Requested facility not      501 Not implemented
      implemented



<span class="grey">Elwell, et al.           Best Current Practice                 [Page 28]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-29" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   70 Only restricted digital     488 Not acceptable here (NOTE 4)
      information available
   79 Service or option not       501 Not implemented
      implemented, unspecified
   87 User not member of CUG      403 Forbidden
   88 Incompatible destination    503 Service unavailable
   102 Recovery on timer expiry   504 Server time-out

   NOTE 3: A QSIG call clearing message containing cause value 16 will
   normally result in the sending of a SIP BYE or CANCEL request.
   However, if a SIP response is to be sent to the INVITE request, the
   default response code should be used.

   NOTE 4: The gateway may include a SIP Warning header if diagnostic
   information in the QSIG Cause information element allows a suitable
   warning code to be selected.

<span class="h4"><a class="selflink" id="section-8.4.2" href="#section-8.4.2">8.4.2</a>.  Receipt of a SIP BYE Request</span>

   On receipt of a SIP BYE request, the gateway SHALL send a QSIG
   DISCONNECT message with cause value 16 (normal call clearing).  The
   gateway SHALL also disconnect media streams, if established, and
   allow QSIG and SIP signalling to complete in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and
   [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], respectively.

   NOTE: When responding to a SIP BYE request, in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>],
   the gateway is also required to respond to any other outstanding
   transactions, e.g., with a SIP 487 (Request Terminated) response.
   This applies in particular if the gateway has not yet returned a
   final response to the SIP INVITE request.

<span class="h4"><a class="selflink" id="section-8.4.3" href="#section-8.4.3">8.4.3</a>.  Receipt of a SIP CANCEL Request</span>

   On receipt of a SIP CANCEL request to clear a call for which the
   gateway has not sent a SIP final response to the received SIP INVITE
   request, the gateway SHALL send a QSIG DISCONNECT message with cause
   value 16 (normal call clearing).  The gateway SHALL also disconnect
   media streams, if established, and allow QSIG and SIP signalling to
   complete in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], respectively.

<span class="h4"><a class="selflink" id="section-8.4.4" href="#section-8.4.4">8.4.4</a>.  Receipt of a SIP 4xx-6xx Response to an INVITE Request</span>

   Except where otherwise specified in the context of overlap sending
   (8.2.2.2), on receipt of a SIP final response (4xx-6xx) to a SIP
   INVITE request, unless the gateway is able to retry the INVITE
   request to avoid the problem (e.g., by supplying authentication in
   the case of a 401 or 407 response), the gateway SHALL transmit a QSIG
   DISCONNECT message.  The cause value in the QSIG DISCONNECT message



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   SHALL be derived from the SIP 4xx-6xx response according to Table 2.
   Cause value 31 (Normal, unspecified) SHALL be used as the default for
   SIP responses not shown in Table 2.  The gateway SHALL also
   disconnect media streams, if established, and allow QSIG and SIP
   signalling to complete in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>] and [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], respectively.

   When generating a QSIG Cause information element, the location field
   SHOULD contain the value "user", if generated as a result of a SIP
   response code 6xx, or the value "Private network serving the remote
   user" in other circumstances.

   Table 2: Mapping of SIP 4xx-6xx responses to an INVITE request to
   QSIG Cause values

   SIP response                        QSIG Cause value (NOTE 6)
   ------------------------------------------------------------------
   400 Bad request                     41  Temporary failure
   401 Unauthorized                    21  Call rejected (NOTE 5)
   402 Payment required                21  Call rejected
   403 Forbidden                       21  Call rejected
   404 Not found                       1   Unallocated number
   405 Method not allowed              63  Service or option
                                           unavailable, unspecified
   406 Not acceptable                  79  Service or option not
                                           implemented, unspecified
   407 Proxy Authentication required   21  Call rejected (NOTE 5)
   408 Request timeout                 102 Recovery on timer expiry
   410 Gone                            22  Number changed
   413 Request entity too large        127 Interworking, unspecified
                                           (NOTE 6)
   414 Request-URI too long            127 Interworking, unspecified
                                           (NOTE 6)
   415 Unsupported media type          79  Service or option not
                                           implemented, unspecified
                                           (NOTE 6)
   416 Unsupported URI scheme          127 Interworking, unspecified
                                           (NOTE 6)
   420 Bad extension                   127 Interworking, unspecified
                                           (NOTE 6)
   421 Extension required              127 Interworking, unspecified
                                           (NOTE 6)
   423 Interval too brief              127 Interworking, unspecified
                                           (NOTE 6)
   480 Temporarily unavailable         18  No user responding
   481 Call/transaction does not exist 41  Temporary failure
   482 Loop detected                   25  Exchange routing error
   483 Too many hops                   25  Exchange routing error




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   484 Address incomplete              28  Invalid number format
                                           (NOTE 6)
   485 Ambiguous                       1   Unallocated Number
   486 Busy here                       17  User busy
   487 Request terminated              (NOTE 7)
   488 Not Acceptable Here             65  Bearer capability not
                                           implemented or 31 Normal,
                                           unspecified (NOTE 8)
   500 Server internal error           41  Temporary failure
   501 Not implemented                 79  Service or option not
                                           implemented, unspecified
   502 Bad gateway                     38  Network out of order
   503 Service unavailable             41  Temporary failure
   504 Gateway time-out                102 Recovery on timer expiry
   505 Version not supported           127 Interworking, unspecified
                                           (NOTE 6)
   513 Message too large               127 Interworking, unspecified
                                           (NOTE 6)
   600 Busy everywhere                 17  User busy
   603 Decline                         21  Call rejected
   604 Does not exist anywhere         1   Unallocated number
   606 Not acceptable                  65  Bearer capability not
                                           implemented or
                                       31  Normal, unspecified (NOTE 8)

   NOTE 5: In some cases, it may be possible for the gateway to provide
   credentials to the SIP UAS that is rejecting an INVITE due to
   authorization failure.  If the gateway can authenticate itself, then
   obviously it should do so and proceed with the call.  Only if the
   gateway cannot authorize itself should the gateway clear the call in
   the QSIG network with this cause value.

   NOTE 6: For some response codes, the gateway may be able to retry the
   INVITE request in order to work around the problem.  In particular,
   this may be the case with response codes indicating a protocol error.
   The gateway SHOULD clear the call in the QSIG network with the
   indicated cause value only if retry is not possible or fails.

   NOTE 7: The circumstances in which SIP response code 487 can be
   expected to arise do not require it to be mapped to a QSIG cause
   code, since the QSIG call will normally already be cleared or in the
   process of clearing.  If QSIG call clearing does, however, need to be
   initiated, the default cause value should be used.

   NOTE 8: When the Warning header is present in a SIP 606 or 488
   message, the warning code should be examined to determine whether it
   is reasonable to generate cause value 65.  This cause value should be
   generated only if there is a chance that a new call attempt with



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   different content in the Bearer capability information element will
   avoid the problem.  In other circumstances, the default cause value
   should be used.

<span class="h4"><a class="selflink" id="section-8.4.5" href="#section-8.4.5">8.4.5</a> Gateway-Initiated Call Clearing</span>

   If the gateway initiates clearing of the QSIG call owing to QSIG
   timer expiry, QSIG protocol error, or use of the QSIG RESTART message
   in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>], the gateway SHALL also initiate clearing of
   the SIP call in accordance with <a href="#section-8.4.1">Section 8.4.1</a>.  If this involves the
   sending of a final response to a SIP INVITE request, the gateway
   SHALL use response code 480 (Temporarily Unavailable) if optional
   QSIG timer T301 has expired or, otherwise, response code 408 (Request
   timeout) or 500 (Server internal error), as appropriate.

   If the gateway initiates clearing of the SIP call owing to SIP timer
   expiry or SIP protocol error in accordance with [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], the gateway
   SHALL also initiate clearing of the QSIG call in accordance with [<a href="#ref-2" title="&quot;Private Integrated Services Network - Circuit-mode Bearer Services - Inter-Exchange Signalling Procedures and Protocol&quot;">2</a>]
   using cause value 102 (Recovery on timer expiry) or 41 (Temporary
   failure), as appropriate.

<span class="h3"><a class="selflink" id="section-8.5" href="#section-8.5">8.5</a>.  Request to Change Media Characteristics</span>

   If after a call has been successfully established the gateway
   receives a SIP INVITE request to change the media characteristics of
   the call in a way that would be incompatible with the bearer
   capability in use within the PISN, the gateway SHALL send back a SIP
   488 (Not Acceptable Here) response and SHALL NOT change the media
   characteristics of the existing call.

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  Number Mapping</span>

   In QSIG, users are identified by numbers, as defined in [<a href="#ref-1" title="&quot;Private Integrated Services Networks (PISN) - Addressing&quot;">1</a>].  Numbers
   are conveyed within the Called party number, Calling party number,
   and Connected number information elements.  The Calling party number
   and Connected number information elements also contain a presentation
   indicator, which can indicate that privacy is required (presentation
   restricted), and a screening indicator, which indicates the source
   and authentication status of the number.

   In SIP, users are identified by Universal Resource Identifiers (URIs)
   conveyed within the Request-URI and various headers, including the
   From and To headers specified in [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>] and optionally the P-Asserted-
   Identity header specified in [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>].  In addition, privacy is indicated
   by the Privacy header specified in [<a href="#ref-13" title="&quot;A Privacy Mechanism for the Session Initiation Protocol (SIP)&quot;">13</a>].






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   This clause specifies the mapping between QSIG Called party number,
   Calling party number, and Connected number information elements and
   corresponding elements in SIP.

   A gateway MAY implement the P-Asserted-Identity header in accordance
   with [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>].  If a gateway implements the P-Asserted-Identity header,
   it SHALL also implement the Privacy header in accordance with [<a href="#ref-13" title="&quot;A Privacy Mechanism for the Session Initiation Protocol (SIP)&quot;">13</a>].
   If a gateway does not implement the P-Asserted-Identity header, it
   MAY implement the Privacy header.

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  Mapping from QSIG to SIP</span>

   The method used to convert a number to a URI is outside the scope of
   this specification.  However, the gateway SHOULD take account of the
   Numbering Plan (NPI) and Type Of Number (TON) fields in the QSIG
   information element concerned when interpreting a number.

   Some aspects of mapping depend on whether the gateway is in the same
   trust domain (as defined in [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>]) as the next hop SIP node (i.e., the
   proxy or UA to which the INVITE request is sent or from which INVITE
   request is received) to honour requests for identity privacy in the
   Privacy header.  This will be network-dependent, and it is
   RECOMMENDED that gateways supporting the P-Asserted-Identity header
   hold a configurable list of next hop nodes that are to be trusted in
   this respect.

<span class="h4"><a class="selflink" id="section-9.1.1" href="#section-9.1.1">9.1.1</a>.  Using Information from the QSIG Called Party Number Information</span>
<span class="h4">        Element</span>

   When mapping a QSIG SETUP message to a SIP INVITE request, the
   gateway SHALL convert the number in the QSIG Called party number
   information to a URI and include that URI in the SIP Request-URI and
   in the To header.

<span class="h4"><a class="selflink" id="section-9.1.2" href="#section-9.1.2">9.1.2</a>.  Using Information from the QSIG Calling Party Number Information</span>
<span class="h4">        Element</span>

   When mapping a QSIG SETUP message to a SIP INVITE request, the
   gateway SHALL use the Calling party number information element, if
   present, as follows.

   If the information element contains a number, the gateway SHALL
   attempt to derive a URI from that number.  Further behaviour depends
   on whether a URI has been derived and the value of the presentation
   indication.






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<span class="h5"><a class="selflink" id="section-9.1.2.1" href="#section-9.1.2.1">9.1.2.1</a>.  No URI derived, and presentation indicator does not have value</span>
          "presentation restricted"

   In this case (including the case where the Calling party number
   information element is absent), the gateway SHALL include a URI
   identifying the gateway in the From header.  Also, if the gateway
   supports the mechanism defined in [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>], the gateway SHALL NOT
   generate a P-Asserted-Identity header.

<span class="h5"><a class="selflink" id="section-9.1.2.2" href="#section-9.1.2.2">9.1.2.2</a>.  No URI derived, and presentation indicator has value</span>
          "presentation restricted"

   In this case, the gateway SHALL generate an anonymous From header.
   Also, if the gateway supports the mechanism defined in [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>], the
   gateway SHALL generate a Privacy header field with parameter
   priv-value = "id" and SHALL NOT generate a P-Asserted-Identity
   header.  The inclusion of additional values of the priv-value
   parameter in the Privacy header is outside the scope of this
   specification.

<span class="h5"><a class="selflink" id="section-9.1.2.3" href="#section-9.1.2.3">9.1.2.3</a>.  URI derived, and presentation indicator has value</span>
          "presentation restricted"

   If the gateway supports the P-Asserted-Identity header and trusts the
   next hop proxy to honour the Privacy header, the gateway SHALL
   generate a P-Asserted-Identity header containing the derived URI,
   SHALL generate a Privacy header with parameter priv-value = "id", and
   SHALL generate an anonymous From header.  The inclusion of additional
   values of the priv-value parameter in the Privacy header is outside
   the scope of this specification.

   If the gateway does not support the P-Asserted-Identity header or
   does not trust the proxy to honour the Privacy header, the gateway
   SHALL behave as in <a href="#section-9.1.2.2">Section 9.1.2.2</a>.

<span class="h5"><a class="selflink" id="section-9.1.2.4" href="#section-9.1.2.4">9.1.2.4</a>.  URI derived, and presentation indicator does not have value</span>
          "presentation restricted"

   In this case, the gateway SHALL generate a P-Asserted-Identity header
   containing the derived URI if the gateway supports this header, SHALL
   NOT generate a Privacy header, and SHALL include the derived URI in
   the From header.  In addition, the gateway MAY use S/MIME, as
   described in Section 23 of [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], to sign a copy of the From header
   included in a message/sipfrag body of the INVITE request as described
   in [<a href="#ref-20" title="&quot;Internet Media Type message/sipfrag&quot;">20</a>].






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<span class="h4"><a class="selflink" id="section-9.1.3" href="#section-9.1.3">9.1.3</a>.  Using Information from the QSIG Connected Number Information</span>
<span class="h4">        Element</span>

   When mapping a QSIG CONNECT message to a SIP 200 (OK) response to an
   INVITE request, the gateway SHALL use the Connected number
   information element, if present, as follows.

   If the information element contains a number, the gateway SHALL
   attempt to derive a URI from that number.  Further behaviour depends
   on whether a URI has been derived and the value of the presentation
   indication.

<span class="h5"><a class="selflink" id="section-9.1.3.1" href="#section-9.1.3.1">9.1.3.1</a>.  No URI derived, and presentation indicator does not have value</span>
          "presentation restricted"

   In this case (including the case where the Connected number
   information element is absent), the gateway SHALL NOT generate a
   P-Asserted-Identity header and SHALL NOT generate a Privacy header.

<span class="h5"><a class="selflink" id="section-9.1.3.2" href="#section-9.1.3.2">9.1.3.2</a>.  No URI derived, and presentation indicator has value</span>
          "presentation restricted"

   In this case, if the gateway supports the mechanism defined in [<a href="#ref-14" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">14</a>],
   the gateway SHALL generate a Privacy header field with parameter
   priv-value = "id" and SHALL NOT generate a P-Asserted-Identity
   header.  The inclusion of additional values of the priv-value
   parameter in the Privacy header is outside the scope of this
   specification.

<span class="h5"><a class="selflink" id="section-9.1.3.3" href="#section-9.1.3.3">9.1.3.3</a>.  URI derived, and presentation indicator has value</span>
          "presentation restricted"

   If the gateway supports the P-Asserted-Identity header and trusts the
   next hop proxy to honour the Privacy header, the gateway SHALL
   generate a P-Asserted-Identity header containing the derived URI and
   SHALL generate a Privacy header with parameter priv-value = "id".
   The inclusion of additional values of the priv-value parameter in the
   Privacy header is outside the scope of this specification.

   If the gateway does not support the P-Asserted-Identity header or
   does not trust the proxy to honour the Privacy header, the gateway
   SHALL behave as in <a href="#section-9.1.3.2">Section 9.1.3.2</a>.









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<span class="h5"><a class="selflink" id="section-9.1.3.4" href="#section-9.1.3.4">9.1.3.4</a>.  URI derived, and presentation indicator does not have value</span>
          "presentation restricted"

   In this case, the gateway SHALL generate a P-Asserted-Identity header
   containing the derived URI if the gateway supports this header and
   SHALL NOT generate a Privacy header.  In addition, the gateway MAY
   use S/MIME, as described in Section 23 of [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>], to sign a To header
   containing the derived URI, the To header being included in a
   message/sipfrag body of the INVITE response as described in [<a href="#ref-20" title="&quot;Internet Media Type message/sipfrag&quot;">20</a>].

   NOTE: The To header in the message/sipfrag body may differ from the
   to header in the response's headers.

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a>.  Mapping from SIP to QSIG</span>

   The method used to convert a URI to a number is outside the scope of
   this specification.  However, NPI and TON fields in the QSIG
   information element concerned SHALL be set to appropriate values in
   accordance with [<a href="#ref-1" title="&quot;Private Integrated Services Networks (PISN) - Addressing&quot;">1</a>].

   Some aspects of mapping depend on whether the gateway trusts the next
   hop SIP node (i.e., the proxy or UA to which the INVITE request is
   sent or from which INVITE request is received) to provide accurate
   information in the P-Asserted-Identity header.  This will be
   network-dependent, and it is RECOMMENDED that gateways hold a
   configurable list of next hop nodes that are to be trusted in this
   respect.

   Some aspects of mapping depend on whether the gateway is prepared to
   use a URI in the From header to derive a number for the Calling party
   number information element.  The default behaviour SHOULD be not to
   use an unsigned or unvalidated From header for this purpose, since in
   principle the information comes from an untrusted source (the remote
   UA).  However, it is recognised that some network administrations may
   believe that the benefits to be derived from supplying a calling
   party number outweigh any risks of supplying false information.
   Therefore, a gateway MAY be configurable to use an unsigned or
   unvalidated From header for this purpose.

<span class="h4"><a class="selflink" id="section-9.2.1" href="#section-9.2.1">9.2.1</a>.  Generating the QSIG Called Party Number Information Element</span>

   When mapping a SIP INVITE request to a QSIG SETUP message, the
   gateway SHALL convert the URI in the SIP Request-URI to a number and
   include that number in the QSIG Called party number information
   element.






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   NOTE: The To header should not be used for this purpose.  This is
   because re-targeting of the request in the SIP network can change the
   Request-URI but leave the To header unchanged.  It is important that
   routing in the QSIG network be based on the final target from the SIP
   network.

<span class="h4"><a class="selflink" id="section-9.2.2" href="#section-9.2.2">9.2.2</a>.  Generating the QSIG Calling Party Number Information Element</span>

   When mapping a SIP INVITE request to a QSIG SETUP message, the
   gateway SHALL generate a Calling party number information element as
   follows.

   If the SIP INVITE request contains an S/MIME signed message/sipfrag
   body [<a href="#ref-20" title="&quot;Internet Media Type message/sipfrag&quot;">20</a>] containing a From header, and if the gateway supports this
   capability and can verify the authenticity and trustworthiness of
   this information, the gateway SHALL attempt to derive a number from
   the URI in that header.  If no number is derived from a
   message/sipfrag body, if the SIP INVITE request contains a P-
   Asserted-Identity header, and if the gateway supports that header and
   trusts the information therein, the gateway SHALL attempt to derive a
   number from the URI in that header.  If a number is derived from one
   of these headers, the gateway SHALL include it in the Calling party
   number information element and include value "network provided" in
   the screening indicator.

   If no number is derivable as described above and if the gateway is
   prepared to use the unsigned or unvalidated From header, the gateway
   SHALL attempt to derive a number from the URI in the From header.  If
   a number is derived from the From header, the gateway SHALL include
   it in the Calling party number information element and include value
   "user provided, not screened" in the screening indicator.

   If no number is derivable, the gateway SHALL NOT include a number in
   the Calling party number information element.

   If the SIP INVITE request contains a Privacy header with value "id"
   in parameter priv-value and the gateway supports this header, or if
   the value in the From header indicates anonymous, the gateway SHALL
   include value "presentation restricted" in the presentation
   indicator.  Based on local policy, the gateway MAY use the presence
   of other priv-values to set the presentation indicator to
   "presentation restricted".  Otherwise the gateway SHALL include value
   "presentation allowed" if a number is present or "not available due
   to interworking" if no number is present.







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   If the resulting Calling party number information element contains no
   number and contains value "not available due to interworking" in the
   presentation indicator, the gateway MAY omit the information element
   from the QSIG SETUP message.

<span class="h4"><a class="selflink" id="section-9.2.3" href="#section-9.2.3">9.2.3</a>.  Generating the QSIG Connected Number Information Element</span>

   When mapping a SIP 2xx response to an INVITE request to a QSIG
   CONNECT message, the gateway SHALL generate a Connected number
   information element as follows.

   If the SIP 2xx response contains an S/MIME signed message/sipfrag
   [<a href="#ref-20" title="&quot;Internet Media Type message/sipfrag&quot;">20</a>] body containing a To header and the gateway supports this
   capability and can verify the authenticity and trustworthiness of
   this information, the gateway SHALL attempt to derive a number from
   the URI in that header.  If no number is derived from a
   message/sipfrag body, if the SIP 2xx response contains a
   P-Asserted-Identity header, and if the gateway supports that header
   and trusts the information therein, the gateway SHALL attempt to
   derive a number from the URI in that header.  If a number is derived
   from one of these headers, the gateway SHALL include it in the
   Connected number information element and include value "network
   provided" in the screening indicator.

   If no number is derivable as described above, the gateway SHOULD NOT
   include a number in the Connected number information element.

   If the SIP 2xx response contains a Privacy header with value "id" in
   parameter priv-value and the gateway supports this header, the
   gateway SHALL include value "presentation restricted" in the
   presentation indicator.  Based on local policy, the gateway MAY use
   the presence of other priv-values to set the presentation indicator
   to "presentation restricted".  Otherwise, the gateway SHALL include
   value "presentation allowed" if a number is present or "not available
   due to interworking" if no number is present.

   If the resulting Connected number information element contains no
   number and value "not available due to interworking" in the
   presentation indicator, the gateway MAY omit the information element
   from the QSIG CONNECT message.











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<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a>.  Requirements for Support of Basic Services</span>

   This document specifies signalling interworking for basic services
   that provide a bi-directional transfer capability for speech,
   facsimile, and modem media between the two networks.

<span class="h3"><a class="selflink" id="section-10.1" href="#section-10.1">10.1</a>.  Derivation of QSIG Bearer Capability Information Element</span>

   The gateway SHALL generate the Bearer Capability Information Element
   in the QSIG SETUP message based on SDP offer information received
   along with the SIP INVITE request.  If the SIP INVITE request does
   not contain SDP offer information or the media type in the SDP offer
   information is only 'audio', then the Bearer capability information
   element SHALL BE generated according to Table 3.  Coding of the
   Bearer capability information element for other media types is
   outside the scope of this specification.

   In addition, the gateway MAY include a Low layer compatibility
   information element and/or High layer compatibility information in
   the QSIG SETUP message if the gateway is able to derive relevant
   information from the SDP offer information.  Specific mappings are
   outside the scope of this specification.

      Table 3: Bearer capability encoding for 'audio' transfer

   Field                          Value
   -----------------------------------------------------------------
   Coding Standard                "CCITT standardized coding" (00)
   Information transfer           "3,1 kHz audio" (10000)
   capability
   Transfer mode                  "circuit mode" (00)
   Information transfer rate      "64 Kbits/s" (10000)
   Multiplier                     Octet omitted
   User information layer 1       Generated by gateway based on
   protocol                       Information of the PISN.  Supported
                                  values are
                                  "CCITT recommendation G.711 mu-law"
                                  (00010)
                                  "CCITT recommendation G.711 A-law"
                                  (00011)

<span class="h3"><a class="selflink" id="section-10.2" href="#section-10.2">10.2</a>.  Derivation of Media Type in SDP</span>

   The gateway SHALL generate SDP offer information to include in the
   SIP INVITE request based on information in the QSIG SETUP message.
   The gateway MAY take account of QSIG Low layer compatibility and/or
   High layer compatibility information elements, if present in the QSIG
   SETUP message, when deriving SDP offer information, in which case



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   specific mappings are outside the scope of this specification.
   Otherwise, the gateway shall generate SDP offer information based
   only on the Bearer capability information element in the QSIG SETUP
   message, in which case the media type SHALL be derived according to
   Table 4.

      Table 4: Media type setting in SDP based on Bearer capability
      information element

   Information transfer capability in          Media type in SDP
   Bearer capability information element
   ---------------------------------------------------------------
   "speech" (00000)                            audio
   "3,1 kHz audio" (10000)                     audio

<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a>.  Security Considerations</span>

<span class="h3"><a class="selflink" id="section-11.1" href="#section-11.1">11.1</a>.  General</span>

   Normal considerations apply for UA use of SIP security measures,
   including digest authentication, TLS, and S/MIME as described in
   [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>].

   The translation of QSIG information elements into SIP headers can
   introduce some privacy and security concerns.  For example, care
   needs to be taken to provide adequate privacy for a user requesting
   presentation restriction if the Calling party number information
   element is openly mapped to the From header.  Procedures for dealing
   with this particular situation are specified in <a href="#section-9.1.2">Section 9.1.2</a>.
   However, since the mapping specified in this document is mainly
   concerned with translating information elements into the headers and
   fields used to route SIP requests, gateways consequently reveal
   (through this translation process) the minimum possible amount of
   information.

   There are some concerns, however, that arise from the other direction
   of mapping, the mapping of SIP headers to QSIG information elements,
   which are enumerated in the following paragraphs.

<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a>.  Calls from QSIG to Invalid or Restricted Numbers</span>

   When end users dial numbers in a PISN, their selections populate the
   Called party number information element in the QSIG SETUP message.
   Similarly, the SIP URI or tel URL and its optional parameters in the
   Request-URI of a SIP INVITE request, which can be created directly by
   end users of a SIP device, map to that information element at a
   gateway.  However, in a PISN, policy can prevent the user from
   dialing certain (invalid or restricted) numbers.  Thus, gateway



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   implementers may wish to provide a means for gateway administrators
   to apply policies restricting the use of certain SIP URIs or tel
   URLs, or SIP URI or tel URL parameters, when authorizing a call from
   SIP to QSIG.

<span class="h3"><a class="selflink" id="section-11.3" href="#section-11.3">11.3</a>.  Abuse of SIP Response Code</span>

   Some additional risks may result from the mapping of SIP response
   codes to QSIG cause values.  SIP user agents could conceivably
   respond to an INVITE request from a gateway with any arbitrary SIP
   response code, and thus they can dictate (within the boundaries of
   the mappings supported by the gateway) the Q.850 cause code that will
   be sent by the gateway in the resulting QSIG call clearing message.
   Generally speaking, the manner in which a call is rejected is
   unlikely to provide any avenue for fraud or denial of service (e.g.,
   by signalling that a call should not be billed, or that the network
   should take critical resources off-line).  However, gateway
   implementers may wish to make provision for gateway administrators to
   modify the response code to cause value mappings to avoid any
   undesirable network-specific behaviour resulting from the mappings
   recommended in <a href="#section-8.4.4">Section 8.4.4</a>.

<span class="h3"><a class="selflink" id="section-11.4" href="#section-11.4">11.4</a>.  Use of the To Header URI</span>

   This specification requires the gateway to map the Request-URI rather
   than the To header in a SIP INVITE request to the Called party number
   information element in a QSIG SETUP message.  Although a SIP UA is
   expected to put the same URI in the To header and in the Request-URI,
   this is not policed by other SIP entities.  Therefore, a To header
   URI that differs from the Request-URI received at the gateway cannot
   be used as a reliable indication that the call has been re-targeted
   in the SIP network or as a reliable indication of the original
   target. Gateway implementers making use of the To header for mapping
   to QSIG elements (e.g., as part of QSIG call diversion signalling)
   may wish to make provision for disabling this mapping when deployed
   in situations where the reliability of the QSIG elements concerned is
   important.

<span class="h3"><a class="selflink" id="section-11.5" href="#section-11.5">11.5</a>.  Use of the From Header URI</span>

   The arbitrary population of the From header of requests by SIP user
   agents has some well-understood security implications for devices
   that rely on the From header as an accurate representation of the
   identity of the originator.  Any gateway that intends to use an
   unsigned or unverified From header to populate the Calling party
   number information element of a QSIG SETUP message should
   authenticate the originator of the request and make sure that it is
   authorized to assert that calling number (or make use of some more



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   secure method to ascertain the identity of the caller).  Note that
   gateways, like all other SIP user agents, MUST support Digest
   authentication as described in [<a href="#ref-10" title="&quot;SIP: Session Initiation Protocol&quot;">10</a>].  Similar considerations apply to
   the use of the SIP P-Asserted-Identity header for mapping to the QSIG
   Calling party number or Connected number information element, i.e.,
   the source of this information should be authenticated.  Use of a
   signed message/sipfrag body to derive a QSIG Calling party number or
   Connected number information element is another secure alternative.

<span class="h3"><a class="selflink" id="section-11.6" href="#section-11.6">11.6</a>.  Abuse of Early Media</span>

   There is another class of potential risk that is related to the cut-
   through of the backwards media path before the call is answered.
   Several practices described in this document involve the connection
   of media streams to user information channels on inter-PINX links and
   the sending of progress description number 1 or 8 in a backward QSIG
   message.  This can result in media being cut through end-to-end, and
   it is possible for the called user agent then to play arbitrary audio
   to the caller for an indefinite period of time before transmitting a
   final response (in the form of a 2xx or higher response code) to an
   INVITE request.  This is useful since it also permits network
   entities (particularly legacy networks that are incapable of
   transmitting Q.850 cause values) to play tones and announcements to
   indicate call failure or call progress, without triggering charging
   by transmitting a 2xx response.  Also, early cut-through can help
   prevent clipping of the initial media when the call is answered.
   There are conceivable respects in which this capability could be used
   fraudulently by the called user agent for transmitting arbitrary
   information without answering the call or before answering the call.
   However, in corporate networks, charging is often not an issue, and
   for calls arriving at a corporate network from a carrier network, the
   carrier network normally takes steps to prevent fraud.

   The usefulness of this capability appears to outweigh any risks
   involved, which may in practice be no greater than in existing
   PISN/ISDN environments.  However, gateway implementers may wish to
   make provision for gateway administrators to turn off cut-through or
   minimise its impact (e.g., by imposing a time limit) when deployed in
   situations where problems can arise.

<span class="h3"><a class="selflink" id="section-11.7" href="#section-11.7">11.7</a>.  Protection from Denial-of-Service Attacks</span>

   Unlike a traditional PISN phone, a SIP user agent can launch multiple
   simultaneous requests in order to reach a particular resource.  It
   would be trivial for a SIP user agent to launch 100 SIP INVITE
   requests at a 100 port gateway, thereby tying up all of its ports.  A
   malicious user could choose to launch requests to telephone numbers
   that are known never to answer, or, where overlap signalling is used,



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   to incomplete addresses.  This could saturate resources at the
   gateway indefinitely, potentially without incurring any charges.
   Gateway implementers may therefore wish to provide means of
   restricting according to policy the number of simultaneous requests
   originating from the same authenticated source, or similar mechanisms
   to address this possible denial-of-service attack.

<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a>.  Acknowledgements</span>

   This document is a product of the authors' activities in Ecma
   (www.ecma-international.org) on interoperability of QSIG with IP
   networks.  An earlier version is published as Standard ECMA-339.
   Ecma has made this work available to the IETF as the basis for
   publishing an RFC.

   The authors wish to acknowledge the assistance of Francois Audet,
   Adam Roach, Jean-Francois Rey, Thomas Stach, and members of Ecma
   TC32-TG17 in preparing and commenting on this document.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a>.  Normative References</span>

   [<a id="ref-1">1</a>]  International Standard ISO/IEC 11571 "Private Integrated
        Services Networks (PISN) - Addressing" (also published by Ecma
        as Standard ECMA-155).

   [<a id="ref-2">2</a>]  International Standard ISO/IEC 11572 "Private Integrated
        Services Network - Circuit-mode Bearer Services - Inter-Exchange
        Signalling Procedures and Protocol" (also published by Ecma as
        Standard ECMA-143).

   [<a id="ref-3">3</a>]  International Standard ISO/IEC 11582 "Private Integrated
        Services Network - Generic Functional Protocol for the Support
        of Supplementary Services - Inter-Exchange Signalling Procedures
        and Protocol" (also published by Ecma as Standard ECMA-165).

   [<a id="ref-4">4</a>]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>, March 1997.

   [<a id="ref-5">5</a>]  Postel, J., "Transmission Control Protocol", STD 7, <a href="./rfc793">RFC 793</a>,
        September 1981.

   [<a id="ref-6">6</a>]  Postel, J., "User Datagram Protocol", STD 6, <a href="./rfc768">RFC 768</a>, August
        1980.

   [<a id="ref-7">7</a>]  Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", <a href="./rfc2246">RFC</a>
        <a href="./rfc2246">2246</a>, January 1999.





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   [<a id="ref-8">8</a>]  Handley, M. and V. Jacobson, "SDP: Session Description
        Protocol", <a href="./rfc2327">RFC 2327</a>, April 1998.

   [<a id="ref-9">9</a>]  Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer,
        H., Taylor, T., Rytina, I., Kalla, M., Zhang, L., and V. Paxson,
        "Stream Control Transmission Protocol", <a href="./rfc2960">RFC 2960</a>, October 2000.

   [<a id="ref-10">10</a>] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
        Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
        Session Initiation Protocol", <a href="./rfc3261">RFC 3261</a>, June 2002.

   [<a id="ref-11">11</a>] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
        Responses in Session Initiation Protocol (SIP)", <a href="./rfc3262">RFC 3262</a>, June
        2002.

   [<a id="ref-12">12</a>] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
        Session Description Protocol (SDP)", <a href="./rfc3264">RFC 3264</a>, June 2002.

   [<a id="ref-13">13</a>] Peterson, J., "A Privacy Mechanism for the Session Initiation
        Protocol (SIP)", <a href="./rfc3323">RFC 3323</a>, November 2002.

   [<a id="ref-14">14</a>] Jennings, C., Peterson, J., and M. Watson, "Private Extensions
        to the Session Initiation Protocol (SIP) for Asserted Identity
        within Trusted Networks", <a href="./rfc3325">RFC 3325</a>, November 2002.

   [<a id="ref-15">15</a>] Postel, J., "Internet Protocol", STD 5, <a href="./rfc791">RFC 791</a>, September 1981.

   [<a id="ref-16">16</a>] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6)
        Specification", <a href="./rfc2460">RFC 2460</a>, December 1998.

   [<a id="ref-17">17</a>] ITU-T Recommendation E.164, "The International Public
        Telecommunication Numbering Plan", (1997-05).

   [<a id="ref-18">18</a>] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Mapping of
        Integrated Services Digital Network (ISDN) User Part (ISUP)
        Overlap Signalling to the Session Initiation Protocol (SIP)",
        <a href="./rfc3578">RFC 3578</a>, August 2003.

   [<a id="ref-19">19</a>] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
        Method", <a href="./rfc3311">RFC 3311</a>, October 2002.

   [<a id="ref-20">20</a>] Sparks, R., "Internet Media Type message/sipfrag", <a href="./rfc3420">RFC 3420</a>,
        November 2002.








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<span class="h2"><a class="selflink" id="appendix-A" href="#appendix-A">Appendix A</a>.  Example Message Sequences</span>

<span class="h3"><a class="selflink" id="appendix-A.1" href="#appendix-A.1">A.1</a>.  Introduction</span>

   This appendix shows some typical message sequences that can occur for
   an interworking between QSIG and SIP.  It is informative.

   NOTE: For all message sequence diagrams, there is no message mapping
   between QSIG and SIP unless explicitly indicated by dotted lines.
   Also, if there are no dotted lines connecting two messages, this
   means that these are independent of each other in terms of the time
   when they occur.

   NOTE: Numbers prefixing SIP method names and response codes in the
   diagrams represent sequence numbers.  Messages bearing the same
   number will have the same value in the CSeq header.

   NOTE: In these examples, SIP provisional responses (other than 100)
   are shown as being sent reliably, using the PRACK method for
   acknowledgement.

<span class="h3"><a class="selflink" id="appendix-A.2" href="#appendix-A.2">A.2</a>.  Message Sequences for Call Establishment from QSIG to SIP</span>

   Below are typical message sequences for successful call establishment
   from QSIG to SIP


























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<span class="h4"><a class="selflink" id="appendix-A.2.1" href="#appendix-A.2.1">A.2.1</a>.  QSIG to SIP, using en bloc procedures on both QSIG and SIP</span>

                           +-------------------+
                           |                   |
                           |     GATEWAY       |
        PISN               |                   |        IP NETWORK
        |                  +-----+------+------+                 |
        |                        |      |                        |
        |                        |      |                        |
        |   QSIG SETUP           |      |        1-INVITE        |
       1|-----------------------&gt;|......|-----------------------&gt;| 2
        |                        |      |                        |
        |                        |      |                        |
        | QSIG CALL PROCEEDING   |      |        1-100 TRYING    |
       3|&lt;-----------------------|      |&lt;-----------------------+ 4
        |                        |      |                        |
        |                        |      |                        |
        |   QSIG ALERTING        |      |        1-180 RINGING   |
       8|&lt;-----------------------|......|&lt;-----------------------+ 5
        |                        |      |                        |
        |                        |      |        2-PRACK         |
        |                        |      |-----------------------&gt;| 6
        |                        |      |        2-200 OK        |
        |                        |      |&lt;-----------------------+ 7
        |                        |      |                        |
        |   QSIG CONNECT         |      |        1-200 OK        |
      11|&lt;-----------------------|......|&lt;-----------------------+ 9
        |                        |      |                        |
        |   QSIG CONNECT ACK     |      |        1-ACK           |
      12|-----------------------&gt;|      |-----------------------&gt;| 10
        |                        |      |                        |
        |&lt;======================&gt;|      |&lt;======================&gt;|
        |        AUDIO           |      |         AUDIO          |

   Figure 3: Typical message sequence for successful call establishment
   from QSIG to SIP, using en bloc procedures on both QSIG and SIP

   1  The PISN sends a QSIG SETUP message to the gateway to begin a
      session with a SIP UA.
   2  On receipt of the QSIG SETUP message, the gateway generates a SIP
      INVITE request and sends it to an appropriate SIP entity in the IP
      network based on the called number.
   3  The gateway sends a QSIG CALL PROCEEDING message to the PISN; no
      more QSIG INFORMATION messages will be accepted.
   4  The IP network sends a SIP 100 (Trying) response to the gateway.
   5  The IP network sends a SIP 180 (Ringing) response.





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   6  The gateway may send back a SIP PRACK request to the IP network
      based on the inclusion of a Require header or a Supported header
      with option tag 100rel in the initial SIP INVITE request.
   7  The IP network sends a SIP 200 (OK) response to the gateway to
      acknowledge the SIP PRACK request
   8  The gateway maps this SIP 180 (Ringing) response to a QSIG
      ALERTING message and sends it to the PISN.
   9  The IP network sends a SIP 200 (OK) response when the call is
      answered.
   10 The gateway sends a SIP ACK request to acknowledge the SIP 200
      (OK) response.
   11 The gateway maps this SIP 200 (OK) response to a QSIG CONNECT
      message and sends it to the PISN.
   12 The PISN sends a QSIG CONNECT ACKNOWLEDGE message in response to
      the QSIG CONNECT message.




































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<span class="h4"><a class="selflink" id="appendix-A.2.2" href="#appendix-A.2.2">A.2.2</a>.  QSIG to SIP, using overlap receiving on QSIG and en bloc sending</span>
        on SIP

                        +------------------------+
     PISN               |         GATEWAY        |      IP NETWORK
                        |                        |
     |  QSIG SETUP      +--------+-------+-------+                |
    1|--------------------------&gt;|       |                        |
     |                           |       |                        |
     |  QSIG SETUP ACK           |       |                        |
    2|&lt;--------------------------|       |                        |
     |                           |       |                        |
     | QSIG INFORMATION          |       |                        |
    3|--------------------------&gt;|       |                        |
     |                           |       |                        |
     | QSIG INFORMATION          |       |  1-INVITE              |
   3a|--------------------------&gt;|.......|-----------------------&gt;|4
     | QSIG CALL PROCEEDING      |       |  1-100 TRYING          |
    5|&lt;--------------------------|       |&lt;-----------------------|6
     |                           |       |                        |
     | QSIG ALERTING             |       |  1-180 RINGING         |
   10|&lt;--------------------------|.......|&lt;-----------------------|7
     |                           |       |  2-PRACK               |
     |                           |       |-----------------------&gt;|8
     |                           |       |  2-200 OK              |
     |                           |       |&lt;-----------------------|9
     | QSIG CONNECT              |       |  1-200 OK              |
   13|&lt;--------------------------|.......|&lt;-----------------------|11
     |                           |       |                        |
     | QSIG CONNECT ACK          |       |  1-ACK                 |
   14|--------------------------&gt;|       |-----------------------&gt;|12
     |          AUDIO            |       |           AUDIO        |
     |&lt;=========================&gt;|       |&lt;======================&gt;|

   Figure 4: Typical message sequence for successful call establishment
   from QSIG to SIP, using overlap receiving on QSIG and en bloc sending
   on SIP

   1  The PISN sends a QSIG SETUP message to the gateway to begin a
      session with a SIP UA.  The QSIG SETUP message does not contain a
      Sending Complete information element.
   2  The gateway sends a QSIG SETUP ACKNOWLEDGE message to the PISN.
      More digits are expected.
   3  More digits are sent from the PISN within a QSIG INFORMATION
      message.
   3a More digits are sent from the PISN within a QSIG INFORMATION
      message.  The QSIG INFORMATION message contains a Sending Complete
      information element.



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   4  The Gateway generates a SIP INVITE request and sends it to an
      appropriate SIP entity in the IP network, based on the called
      number.
   5  The gateway sends a QSIG CALL PROCEEDING message to the PISN; no
      more QSIG INFORMATION messages will be accepted.
   6  The IP network sends a SIP 100 (Trying) response to the gateway.
   7  The IP network sends a SIP 180 (Ringing) response.
   8  The gateway may send back a SIP PRACK request to the IP network
      based on the inclusion of a Require header or a Supported header
      with option tag 100rel in the initial SIP INVITE request.
   9  The IP network sends a SIP 200 (OK) response to the gateway to
      acknowledge the SIP PRACK request.
   10 The gateway maps this SIP 180 (Ringing) response to a QSIG
      ALERTING message and sends it to the PINX.
   11 The IP network sends a SIP 200 (OK) response when the call is
      answered.
   12 The gateway sends an SIP ACK request to acknowledge the SIP 200
      (OK) response.
   13 The gateway maps this SIP 200 (OK) response to a QSIG CONNECT
      message and sends it to the PINX.
   14 The PISN sends a QSIG CONNECT ACKNOWLEDGE message in response to
      the QSIG CONNECT message.





























<span class="grey">Elwell, et al.           Best Current Practice                 [Page 49]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-50" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.2.3" href="#appendix-A.2.3">A.2.3</a>.  QSIG to SIP, using overlap procedures on both QSIG and SIP</span>

                        +----------------------+
     PISN               |        GATEWAY       |         IP NETWORK
                        |                      |
     |  QSIG SETUP      +-------+-------+------+                  |
   1 |-------------------------&gt;|       |                         |
     |                          |       |                         |
     |  QSIG SETUP ACK          |       |                         |
   2 |&lt;-------------------------|       |                         |
     |                          |       |                         |
     | QSIG INFORMATION         |       |                         |
   3 |-------------------------&gt;|       |                         |
     | QSIG INFORMATION         |       | 1-INVITE                |
   <a href="#section-3">3</a> |-------------------------&gt;|.......|------------------------&gt;|4
     |                          |       | 1-484                   |
     |                          |       |&lt;------------------------|5
     |                          |       | 1-ACK                   |
     |                          |       |------------------------&gt;|6
     | QSIG INFORMATION         |       | 2-INVITE                |
   <a href="#section-7">7</a> |-------------------------&gt;|.......|------------------------&gt;|4
     |                          |       | 2-484                   |
     |                          |       |&lt;------------------------|5
     |                          |       | 2-ACK                   |
     |                          |       |------------------------&gt;|6
     |                          |       |                         |
     | QSIG INFORMATION         |       |                         |
     | Sending Complete IE      |       | 3-INVITE                |
   <a href="#section-8">8</a> |-------------------------&gt;|.......|------------------------&gt;|10
     | QSIG CALL PROCEEDING     |       | 3-100 TRYING            |
   9 |&lt;-------------------------|       |&lt;------------------------|11
     |                          |       |                         |
     | QSIG ALERTING            |       | 3-180 RINGING           |
   15|&lt;-------------------------|.......|&lt;------------------------|12
     |                          |       | 4-PRACK                 |
     |                          |       |------------------------&gt;|13
     |                          |       | 4-200 OK                |
     |                          |       |&lt;------------------------|14
     | QSIG CONNECT             |       | 3-200 OK                |
   18|&lt;-------------------------|.......|&lt;------------------------|16
     |                          |       |                         |
     | QSIG CONNECT ACK         |       | 3-ACK                   |
   19|-------------------------&gt;|       |------------------------&gt;|17
     |         AUDIO            |       |         AUDIO           |
     |&lt;========================&gt;|       |&lt;=======================&gt;|
     |                          |       |                         |





<span class="grey">Elwell, et al.           Best Current Practice                 [Page 50]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-51" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   Figure 5: Typical message sequence for successful call establishment
   from QSIG to SIP, using overlap procedures on both QSIG and SIP

   1  The PISN sends a QSIG SETUP message to the gateway to begin a
      session with a SIP UA.  The QSIG SETUP message does not contain a
      Sending complete information element.
   2  The gateway sends a QSIG SETUP ACKNOWLEDGE message to the PISN.
      More digits are expected.
   3  More digits are sent from the PISN within a QSIG INFORMATION
      message.
   4  When the gateway receives the minimum number of digits required to
      route the call, it generates a SIP INVITE request and sends it to
      an appropriate SIP entity in the IP network based on the called
      number
   5  Due to an insufficient number of digits, the IP network will
      return a SIP 484 (Address Incomplete) response.
   6  The SIP 484 (Address Incomplete) response is acknowledged.
   7  More digits are received from the PISN in a QSIG INFORMATION
      message.  A new INVITE is sent with the same Call-ID and From
      values but an updated Request-URI.
   8  More digits are received from the PISN in a QSIG INFORMATION
      message.  The QSIG INFORMATION message contains a Sending Complete
      information element.
   9  The gateway sends a QSIG CALL PROCEEDING message to the PISN; no
      more information will be accepted.
   10 The gateway sends a new SIP INVITE request with an updated
      Request-URI field.
   11 The IP network sends a SIP 100 (Trying) response to the gateway.
   12 The IP network sends a SIP 180 (Ringing) response.
   13 The gateway may send back a SIP PRACK request to the IP network
      based on the inclusion of a Require header or a Supported header
      with option tag 100rel in the initial SIP INVITE request.
   14 The IP network sends a SIP 200 (OK) response to the gateway to
      acknowledge the SIP PRACK request.
   15 The gateway maps this SIP 180 (Ringing) response to a QSIG
      ALERTING message and sends it to the PISN.
   16 The IP network sends a SIP 200 (OK) response when the call is
      answered.
   17 The gateway sends a SIP ACK request to acknowledge the SIP 200
      (OK) response.
   18 The gateway maps this SIP 200 (OK) response to a QSIG CONNECT
      message.
   19 The PISN sends a QSIG CONNECT ACKNOWLEDGE message in response to
      the QSIG CONNECT message.







<span class="grey">Elwell, et al.           Best Current Practice                 [Page 51]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-52" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h3"><a class="selflink" id="appendix-A.3" href="#appendix-A.3">A.3</a>.  Message sequences for call establishment from SIP to QSIG</span>

   Below are typical message sequences for successful call establishment
   from SIP to QSIG

<span class="h4"><a class="selflink" id="appendix-A.3.1" href="#appendix-A.3.1">A.3.1</a>.  SIP to QSIG, using en bloc procedures</span>

                        +----------------------+
     IP NETWORK         |        GATEWAY       |              PISN
                        |                      |
     |                  +-------+-------+------+                  |
     |                          |       |                         |
     |                          |       |                         |
     |     1-INVITE             |       | QSIG SETUP              |
   <a href="#section-1">1</a> |-------------------------&gt;|.......|------------------------&gt;|3
     |     1-100 TRYING         |       | QSIG CALL PROCEEDING    |
   2 |&lt;-------------------------|       |&lt;------------------------|4
     |     1-180 RINGING        |       | QSIG ALERTING           |
   <a href="#section-6">6</a> |&lt;-------------------------|.......|&lt;------------------------|5
     |                          |       |                         |
     |                          |       |                         |
     |     2-PRACK              |       |                         |
   7 |-------------------------&gt;|       |                         |
     |     2-200 OK             |       |                         |
   8 |&lt;-------------------------|       |                         |
     |     1-200 OK             |       | QSIG CONNECT            |
   11|&lt;-------------------------|.......|&lt;------------------------|9
     |                          |       |                         |
     |     1-ACK                |       | QSIG CONNECT ACK        |
   12|-------------------------&gt;|       |------------------------&gt;|10
     |         AUDIO            |       |         AUDIO           |
     |&lt;========================&gt;|       |&lt;=======================&gt;|
     |                          |       |                         |

   Figure 6: Typical message sequence for successful call establishment
   from SIP to QSIG, using en bloc procedures

   1  The IP network sends a SIP INVITE request to the gateway.
   2  The gateway sends a SIP 100 (Trying) response to the IP network.
   3  On receipt of the SIP INVITE request, the gateway sends a QSIG
      SETUP message.
   4  The PISN sends a QSIG CALL PROCEEDING message to the gateway.
   5  A QSIG ALERTING message is returned to indicate that the end user
      in the PISN is being alerted.
   6  The gateway maps the QSIG ALERTING message to a SIP 180 (Ringing)
      response.





<span class="grey">Elwell, et al.           Best Current Practice                 [Page 52]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-53" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   7  The IP network can send back a SIP PRACK request to the IP network
      based on the inclusion of a Require header or a Supported header
      with option tag 100rel in the initial SIP INVITE request.
   8  The gateway sends a SIP 200 (OK) response to acknowledge the SIP
      PRACK request.
   9  The PISN sends a QSIG CONNECT message to the gateway when the call
      is answered.
   10 The gateway sends a QSIG CONNECT ACKNOWLEDGE message to
      acknowledge the QSIG CONNECT message.
   11 The QSIG CONNECT message is mapped to a SIP 200 (OK) response.
   12 The IP network, upon receiving a SIP INVITE final response (200),
      will send a SIP ACK request to acknowledge receipt.







































<span class="grey">Elwell, et al.           Best Current Practice                 [Page 53]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-54" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.3.2" href="#appendix-A.3.2">A.3.2</a>.  SIP to QSIG, using overlap receiving on SIP and en bloc sending</span>
        on QSIG

                        +----------------------+
     IP NETWORK         |        GATEWAY       |               PISN
                        |                      |
     | 1-INVITE         +-------+-------+------+                  |
   1 |-------------------------&gt;|       |                         |
     |     1-484                |       |                         |
   2 |&lt;-------------------------|       |                         |
     |     1-ACK                |       |                         |
   3 |-------------------------&gt;|       |                         |
     |     2-INVITE             |       |                         |
   1 |-------------------------&gt;|       |                         |
     |     2-484                |       |                         |
   2 |&lt;-------------------------|       |                         |
     |     2- ACK               |       |                         |
   3 |-------------------------&gt;|       |                         |
     |     3-INVITE             |       | QSIG SETUP              |
   <a href="#section-4">4</a> |-------------------------&gt;|.......|------------------------&gt;|6
     |     3-100 TRYING         |       | QSIG CALL PROCEEDING    |
   5 |&lt;-------------------------|       |&lt;------------------------|7
     |     3-180 RINGING        |       | QSIG ALERTING           |
   <a href="#section-9">9</a> |&lt;-------------------------|.......|&lt;------------------------|8
     |                          |       |                         |
     |                          |       |                         |
     |     4-PRACK              |       |                         |
   10|-------------------------&gt;|       |                         |
     |     4-200 OK             |       |                         |
   11|&lt;-------------------------|       |                         |
     |     3-200 OK             |       | QSIG CONNECT            |
   14|&lt;-------------------------|.......|&lt;------------------------|12
     |                          |       |                         |
     |     3-ACK                |       | QSIG CONNECT ACK        |
   15|-------------------------&gt;|       |------------------------&gt;|13
     |         AUDIO            |       |         AUDIO           |
     |&lt;========================&gt;|       |&lt;=======================&gt;|
     |                          |       |                         |

   Figure 7: Typical message sequence for successful call establishment
   from SIP to QSIG, using overlap receiving on SIP and en bloc sending
   on QSIG

   1  The IP network sends a SIP INVITE request to the gateway.
   2  Due to an insufficient number of digits, the gateway returns a SIP
      484 (Address Incomplete) response.
   3  The IP network acknowledges the SIP 484 (Address Incomplete)
      response.



<span class="grey">Elwell, et al.           Best Current Practice                 [Page 54]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-55" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   4  The IP network sends a new SIP INVITE request with the same Call-
      ID and updated Request-URI.
   5  The gateway now has all the digits required to route the call to
      the PISN.  The gateway sends back a SIP 100 (Trying) response.
   6  The gateway sends a QSIG SETUP message.
   7  The PISN sends a QSIG CALL PROCEEDING message to the gateway.
   8  A QSIG ALERTING message is returned to indicate that the end user
      in the PISN is being alerted.
   9  The gateway maps the QSIG ALERTING message to a SIP 180 (Ringing)
      response.
   10 The IP network can send back a SIP PRACK request to the IP network
      based on the inclusion of a Require header or a Supported header
      with option tag 100rel in the initial SIP INVITE request.
   11 The gateway sends a SIP 200 (OK) response to acknowledge the SIP
      PRACK request.
   12 The PISN sends a QSIG CONNECT message to the gateway when the call
      is answered.
   13 The gateway sends a QSIG CONNECT ACKNOWLEDGE message to
      acknowledge the CONNECT message.
   14 The QSIG CONNECT message is mapped to a SIP 200 (OK) response.
   15 The IP network, upon receiving a SIP INVITE final response (200),
      will send a SIP ACK request to acknowledge receipt.





























<span class="grey">Elwell, et al.           Best Current Practice                 [Page 55]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-56" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.3.3" href="#appendix-A.3.3">A.3.3</a>.  SIP to QSIG, using overlap procedures on both SIP and QSIG</span>

                        +----------------------+
     IP NETWORK         |        GATEWAY       |               PISN
                        |                      |
     | 1-INVITE         +-------+-------+------+                  |
   1 |-------------------------&gt;|       |                         |
     |     1-484                |       |                         |
   2 |&lt;-------------------------|       |                         |
     |     1-ACK                |       |                         |
   3 |-------------------------&gt;|       |                         |
     |     2-INVITE             |       | QSIG SETUP              |
   <a href="#section-4">4</a> |-------------------------&gt;|.......|------------------------&gt;|6
     |     2-100 TRYING         |       | QSIG SETUP ACK          |
   5 |&lt;-------------------------|       |&lt;------------------------|7
     |     3- INVITE            |       | QSIG INFORMATION        |
   <a href="#section-8">8</a> |-------------------------&gt;|.......|------------------------&gt;|10
     |     3-100 TRYING         |       |                         |
   9 |&lt;-------------------------|       | QSIG CALL PROCEEDING    |
     |                          |       |&lt;------------------------|11
   13|     3-180 RINGING        |       | QSIG ALERTING           |
     |&lt;-------------------------|.......|&lt;------------------------|12
     |     2-484                |       |                         |
   14|&lt;-------------------------|       |                         |
     |     2-ACK                |       |                         |
   15|-------------------------&gt;|       |                         |
     |     4-PRACK              |       |                         |
   16|-------------------------&gt;|       |                         |
     |     4-200 OK             |       |                         |
   17|&lt;-------------------------|       |                         |
     |     3-200 OK             |       | QSIG CONNECT            |
   20|&lt;-------------------------|.......|&lt;------------------------|18
     |                          |       |                         |
     |     3-ACK                |       | QSIG CONNECT ACK        |
   21|-------------------------&gt;|       |------------------------&gt;|19
     |         AUDIO            |       |         AUDIO           |
     |&lt;========================&gt;|       |&lt;=======================&gt;|
     |                          |       |                         |

   Figure 8: Typical message sequence for successful call establishment
   from SIP to QSIG, using overlap procedures on both SIP and QSIG

   1  The IP network sends a SIP INVITE request to the gateway.
   2  Due to an insufficient number of digits, the gateway returns a SIP
      484 (Address Incomplete) response.
   3  The IP network acknowledges the SIP 484 (Address Incomplete)
      response.




<span class="grey">Elwell, et al.           Best Current Practice                 [Page 56]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-57" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


   4  The IP network sends a new SIP INVITE request with the same
      Call-ID and updated Request-URI.
   5  The gateway now has all the digits required to route the call to
      the PISN.  The gateway sends back a SIP 100 (Trying) response to
      the IP network.
   6  The gateway sends a QSIG SETUP message.
   7  The PISN needs more digits to route the call and sends a QSIG
      SETUP ACKNOWLEDGE message to the gateway.
   8  The IP network sends a new SIP INVITE request with the same
      Call-ID and From values and updated Request-URI.
   9  The gateway sends back a SIP 100 (Trying) response to the IP
      network.
   10 The gateway maps the new SIP INVITE request to a QSIG INFORMATION
      message.
   11 The PISN has all the digits required and sends back a QSIG CALL
      PROCEEDING message to the gateway.
   12 A QSIG ALERTING message is returned to indicate that the end user
      in the PISN is being alerted.
   13 The gateway maps the QSIG ALERTING message to a SIP 180 (Ringing)
      response.
   14 The gateway sends a SIP 484 (Address Incomplete) response for the
      previous SIP INVITE request.
   15 The IP network acknowledges the SIP 484 (Address Incomplete)
      response.
   16 The IP network can send back a SIP PRACK request to the IP network
      based on the inclusion of a Require header or a Supported header
      with option tag 100rel in the initial SIP INVITE request.
   17 The gateway sends a SIP 200 (OK) response to acknowledge the SIP
      PRACK request.
   18 The PISN sends a QSIG CONNECT message to the gateway when the call
      is answered.
   19 The gateway sends a QSIG CONNECT ACKNOWLEDGE message to
      acknowledge the QSIG CONNECT message.
   20 The QSIG CONNECT message is mapped to a SIP 200 (OK) response.
   21 The IP network, upon receiving a SIP INVITE final response (200),
      will send a SIP ACK request to acknowledge receipt.















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 57]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-58" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h3"><a class="selflink" id="appendix-A.4" href="#appendix-A.4">A.4</a>.  Message Sequence for Call Clearing from QSIG to SIP</span>

   Below are typical message sequences for Call Clearing from QSIG to
   SIP

<span class="h4"><a class="selflink" id="appendix-A.4.1" href="#appendix-A.4.1">A.4.1</a>.  QSIG to SIP, subsequent to call establishment</span>

                         +-------------------+
                         |                   |
                         |     GATEWAY       |
     PISN                |                   |         IP NETWORK
      |                  +-----+------+------+                 |
      |                        |      |                        |
      |                        |      |                        |
      |     QSIG DISCONNECT    |      |   2- BYE               |
     1|-----------------------&gt;|......|-----------------------&gt;|4
      |     QSIG RELEASE       |      |        2-200 OK        |
     2|&lt;-----------------------|      |&lt;-----------------------|5
      |     QSIG RELEASE COMP  |      |                        |
     3|-----------------------&gt;|      |                        |
      |                        |      |                        |
      |                        |      |                        |
      |                        |      |                        |

   Figure 9: Typical message sequence for call clearing from QSIG to
   SIP, subsequent to call establishment

   1  The PISN sends a QSIG DISCONNECT message to the gateway.
   2  The gateway sends back a QSIG RELEASE message to the PISN in
      response to the QSIG DISCONNECT message.
   3  The PISN sends a QSIG RELEASE COMPLETE message in response.  All
      PISN resources are now released.
   4  The gateway maps the QSIG DISCONNECT message to a SIP BYE request.
   5  The IP network sends back a SIP 200 (OK) response to the SIP BYE
      request.  All IP resources are now released.
















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 58]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-59" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.4.2" href="#appendix-A.4.2">A.4.2</a>.  QSIG to SIP, during establishment of a call from SIP to QSIG</span>

                              +-------------------+
                              |                   |
                              |     GATEWAY       |
           PISN               |                   |       IP NETWORK
           |                  +-----+------+------+                |
           |                        |      |                       |
           |                        |      |                       |
           |     QSIG DISCONNECT    |      |   1- 4XX / 6XX        |
          1|-----------------------&gt;|......|----------------------&gt;|4
           |     QSIG RELEASE       |      |        1- ACK         |
          2|&lt;-----------------------|      |&lt;----------------------|5
           |     QSIG RELEASE COMP  |      |                       |
          3|-----------------------&gt;|      |                       |
           |                        |      |                       |
           |                        |      |                       |

   Figure 10: Typical message sequence for call clearing from QSIG to
   SIP, during establishment of a call from SIP to QSIG (gateway has
   not sent a final response to the SIP INVITE request)

   1  The PISN sends a QSIG DISCONNECT message to the gateway
   2  The gateway sends back a QSIG RELEASE message to the PISN in
      response to the QSIG DISCONNECT message
   3  The PISN sends a QSIG RELEASE COMPLETE message in response.  All
      PISN resources are now released.
   4  The gateway maps the QSIG DISCONNECT message to a SIP 4xx-6xx
      response
   5  The IP network sends back a SIP ACK request in response to the SIP
      4xx-6xx response.  All IP resources are now released




















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 59]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-60" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.4.3" href="#appendix-A.4.3">A.4.3</a>.  QSIG to SIP, during establishment of a call from QSIG to SIP</span>

                             +-------------------+
                             |                   |
                             |     GATEWAY       |
         PISN                |                   |         IP NETWORK
          |                  +-----+------+------+                 |
          |                        |      |                        |
          |                        |      |                        |
          |     QSIG DISCONNECT    |      |   1- CANCEL            |
         1|-----------------------&gt;|......|-----------------------&gt;|4
          |     QSIG RELEASE       |      |1-487 Request Terminated|
         2|&lt;-----------------------|      |&lt;-----------------------|5
          |     QSIG RELEASE COMP  |      |                        |
         3|-----------------------&gt;|      |   1- ACK               |
          |                        |      |-----------------------&gt;|6
          |                        |      |                        |
          |                        |      |   1- 200 OK            |
          |                        |      |&lt;-----------------------|7
          |                        |      |                        |

   Figure 11: Typical message sequence for call clearing from QSIG to
   SIP, during establishment of a call from QSIG to SIP (gateway has
   received a provisional response to the SIP INVITE request but not a
   final response)

   1  The PISN sends a QSIG DISCONNECT message to the gateway.
   2  The gateway sends back a QSIG RELEASE message to the PISN in
      response to the QSIG DISCONNECT message.
   3  The PISN sends a QSIG RELEASE COMPLETE message in response.  All
      PISN resources are now released.
   4  The gateway maps the QSIG DISCONNECT message to a SIP CANCEL
      request (subject to receipt of a provisional response, but not of
      a final response).
   5  The IP network sends back a SIP 487 (Request Terminated) response
      to the SIP INVITE request.
   6  The gateway, on receiving a SIP final response (487) to the SIP
      INVITE request, sends back a SIP ACK request to acknowledge
      receipt.
   7  The IP network sends back a SIP 200 (OK) response to the SIP
      CANCEL request.  All IP resources are now released.










<span class="grey">Elwell, et al.           Best Current Practice                 [Page 60]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-61" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h3"><a class="selflink" id="appendix-A.5" href="#appendix-A.5">A.5</a>.  Message Sequence for Call Clearing from SIP to QSIG</span>

   Below are typical message sequences for Call Clearing from SIP to
   QSIG

<span class="h4"><a class="selflink" id="appendix-A.5.1" href="#appendix-A.5.1">A.5.1</a>.  SIP to QSIG, subsequent to call establishment</span>

                             +-------------------+
                             |                   |
                             |     GATEWAY       |
          IP NETWORK         |                   |              PISN
          |                  +-----+------+------+                 |
          |                        |      |                        |
          |                        |      |                        |
          |   2- BYE               |      |     QSIG DISCONNECT    |
         1|-----------------------&gt;|......|-----------------------&gt;|3
          |                        |      |     QSIG RELEASE       |
          |                        |      |&lt;-----------------------|4
          |        2-200 OK        |      |     QSIG RELEASE COMP  |
         2|&lt;-----------------------|      |-----------------------&gt;|5
          |                        |      |                        |
          |                        |      |                        |

   Figure 12: Typical message sequence for call clearing from SIP to
   QSIG, subsequent to call establishment

   1  The IP network sends a SIP BYE request to the gateway.
   2  The gateway sends back a SIP 200 (OK) response to the SIP BYE
      request.  All IP resources are now released.
   3  The gateway maps the SIP BYE request to a QSIG DISCONNECT message.
   4  The PISN sends back a QSIG RELEASE message to the gateway in
      response to the QSIG DISCONNECT message.
   5  The gateway sends a QSIG RELEASE COMPLETE message in response.
      All PISN resources are now released.

















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 61]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-62" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.5.2" href="#appendix-A.5.2">A.5.2</a>.  SIP to QSIG, during establishment of a call from QSIG to SIP</span>

                             +-------------------+
                             |                   |
                             |     GATEWAY       |
          IP NETWORK         |                   |              PISN
          |                  +-----+------+------+                 |
          |                        |      |                        |
          |                        |      |                        |
          |   1- 4XX / 6XX         |      |     QSIG DISCONNECT    |
         1|-----------------------&gt;|......|-----------------------&gt;|3
          |                        |      |     QSIG RELEASE       |
          |                        |      |&lt;-----------------------|4
          |        1- ACK          |      |     QSIG RELEASE COMP  |
         2|&lt;-----------------------|      |-----------------------&gt;|5
          |                        |      |                        |
          |                        |      |                        |
          |                        |      |                        |

   Figure 13: Typical message sequence for call clearing from SIP to
   QSIG, during establishment of a call from QSIG to SIP (gateway has
   not previously received a final response to the SIP INVITE request)

   1  The IP network sends a SIP 4xx-6xx response to the gateway.
   2  The gateway sends back a SIP ACK request in response to the SIP
      4xx-6xx response.  All IP resources are now released.
   3  The gateway maps the SIP 4xx-6xx response to a QSIG DISCONNECT
      message.
   4  The PISN sends back a QSIG RELEASE message to the gateway in
      response to the QSIG DISCONNECT message.
   5  The gateway sends a QSIG RELEASE COMPLETE message in response.
      All PISN resources are now released.



















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 62]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-63" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


<span class="h4"><a class="selflink" id="appendix-A.5.3" href="#appendix-A.5.3">A.5.3</a>.  SIP to QSIG, during establishment of a call from SIP to QSIG</span>

                             +-------------------+
                             |                   |
                             |     GATEWAY       |
         IP NETWORK          |                   |              PISN
          |                  +-----+------+------+                 |
          |                        |      |                        |
          |                        |      |                        |
          |   1- CANCEL            |      |     QSIG DISCONNECT    |
         1|-----------------------&gt;|......|-----------------------&gt;|4
          |                        |      |     QSIG RELEASE       |
          |                        |      |&lt;-----------------------|5
          |1-487 Request Terminated|      |     QSIG RELEASE COMP  |
         2|&lt;-----------------------|      |-----------------------&gt;|6
          |                        |      |                        |
          |   1- ACK               |      |                        |
         3|-----------------------&gt;|      |                        |
          |                        |      |                        |
          |   1- 200 OK            |      |                        |
         4|&lt;-----------------------|      |                        |

   Figure 14: Typical message sequence for call clearing from SIP to
   QSIG, during establishment of a call from SIP to QSIG (gateway has
   sent a provisional response to the SIP INVITE request but not a final
   response)

   1  The IP network sends a SIP CANCEL request to the gateway.
   2  The gateway sends back a SIP 487 (Request Terminated) response to
      the SIP INVITE request.
   3  The IP network, on receiving a SIP final response (487) to the SIP
      INVITE request, sends back a SIP ACK request to acknowledge
      receipt.
   4  The gateway sends back a SIP 200 (OK) response to the SIP CANCEL
      request.  All IP resources are now released.
   5  The gateway maps the SIP 4xx-6xx response to a QSIG DISCONNECT
      message.
   6  The PISN sends back a QSIG RELEASE message to the gateway in
      response to the QSIG DISCONNECT message.
   7  The gateway sends a QSIG RELEASE COMPLETE message in response.
      All PISN resources are now released.










<span class="grey">Elwell, et al.           Best Current Practice                 [Page 63]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-64" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


Authors' Addresses

   John Elwell
   Siemens plc
   Technology Drive
   Beeston
   Nottingham, UK, NG9 1LA

   EMail: john.elwell@siemens.com


   Frank Derks
   NEC Philips Unified Solutions
   Anton Philipsweg 1
   1223 KZ Hilversum
   The Netherlands

   EMail: frank.derks@nec-philips.com


   Olivier Rousseau
   Alcatel Business Systems
   32,Avenue Kleber
   92700 Colombes
   France

   EMail: Olivier.Rousseau@alcatel.fr


   Patrick Mourot
   Alcatel Business Systems
   1,Rue Dr A.  Schweitzer
   67400 Illkirch
   France

   EMail: Patrick.Mourot@alcatel.fr















<span class="grey">Elwell, et al.           Best Current Practice                 [Page 64]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-65" ></span>
<span class="grey"><a href="./rfc4497">RFC 4497</a>           Interworking between SIP and QSIG            May 2006</span>


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Elwell, et al.           Best Current Practice                 [Page 65]
</pre>