<pre>Network Working Group                                           D. Burke
Request for Comments: 5552                                        Google
Category: Standards Track                                       M. Scott
                                                                 Genesys
                                                                May 2009


                <span class="h1">SIP Interface to VoiceXML Media Services</span>

Status of This Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (c) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
   Provisions Relating to IETF Documents in effect on the date of
   publication of this document (<a href="http://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>).
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.

Abstract

   This document describes a SIP interface to VoiceXML media services.
   Commonly, Application Servers controlling Media Servers use this
   protocol for pure VoiceXML processing capabilities.  This protocol is
   an adjunct to the full MEDIACTRL protocol and packages mechanism.

















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Table of Contents

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-3">3</a>
      <a href="#section-1.1">1.1</a>. Use Cases ..................................................<a href="#page-3">3</a>
           <a href="#section-1.1.1">1.1.1</a>. IVR Services with Application Servers ...............<a href="#page-3">3</a>
           <a href="#section-1.1.2">1.1.2</a>. PSTN IVR Service Node ...............................<a href="#page-4">4</a>
           <a href="#section-1.1.3">1.1.3</a>. 3GPP IMS Media Resource Function (MRF) ..............<a href="#page-5">5</a>
           <a href="#section-1.1.4">1.1.4</a>. CCXML &lt;-&gt; VoiceXML Interaction ......................<a href="#page-6">6</a>
           <a href="#section-1.1.5">1.1.5</a>. Other Use Cases .....................................<a href="#page-6">6</a>
      <a href="#section-1.2">1.2</a>. Terminology ................................................<a href="#page-7">7</a>
   <a href="#section-2">2</a>. VoiceXML Session Establishment and Termination ..................<a href="#page-7">7</a>
      <a href="#section-2.1">2.1</a>. Service Identification .....................................<a href="#page-7">7</a>
      <a href="#section-2.2">2.2</a>. Initiating a VoiceXML Session .............................<a href="#page-10">10</a>
      <a href="#section-2.3">2.3</a>. Preparing a VoiceXML Session ..............................<a href="#page-11">11</a>
      <a href="#section-2.4">2.4</a>. Session Variable Mappings .................................<a href="#page-12">12</a>
      <a href="#section-2.5">2.5</a>. Terminating a VoiceXML Session ............................<a href="#page-15">15</a>
      <a href="#section-2.6">2.6</a>. Examples ..................................................<a href="#page-16">16</a>
           <a href="#section-2.6.1">2.6.1</a>. Basic Session Establishment ........................<a href="#page-16">16</a>
           <a href="#section-2.6.2">2.6.2</a>. VoiceXML Session Preparation .......................<a href="#page-17">17</a>
           <a href="#section-2.6.3">2.6.3</a>. MRCP Establishment .................................<a href="#page-18">18</a>
   <a href="#section-3">3</a>. Media Support ..................................................<a href="#page-19">19</a>
      <a href="#section-3.1">3.1</a>. Offer/Answer ..............................................<a href="#page-19">19</a>
      <a href="#section-3.2">3.2</a>. Early Media ...............................................<a href="#page-19">19</a>
      <a href="#section-3.3">3.3</a>. Modifying the Media Session ...............................<a href="#page-21">21</a>
      <a href="#section-3.4">3.4</a>. Audio and Video Codecs ....................................<a href="#page-21">21</a>
      <a href="#section-3.5">3.5</a>. DTMF ......................................................<a href="#page-22">22</a>
   <a href="#section-4">4</a>. Returning Data to the Application Server .......................<a href="#page-22">22</a>
      <a href="#section-4.1">4.1</a>. HTTP Mechanism ............................................<a href="#page-22">22</a>
      <a href="#section-4.2">4.2</a>. SIP Mechanism .............................................<a href="#page-23">23</a>
   <a href="#section-5">5</a>. Outbound Calling ...............................................<a href="#page-25">25</a>
   <a href="#section-6">6</a>. Call Transfer ..................................................<a href="#page-25">25</a>
      <a href="#section-6.1">6.1</a>. Blind .....................................................<a href="#page-26">26</a>
      <a href="#section-6.2">6.2</a>. Bridge ....................................................<a href="#page-27">27</a>
      <a href="#section-6.3">6.3</a>. Consultation ..............................................<a href="#page-29">29</a>
   <a href="#section-7">7</a>. Contributors ...................................................<a href="#page-31">31</a>
   <a href="#section-8">8</a>. Acknowledgements ...............................................<a href="#page-31">31</a>
   <a href="#section-9">9</a>. Security Considerations ........................................<a href="#page-31">31</a>
   <a href="#section-10">10</a>. IANA Considerations ...........................................<a href="#page-32">32</a>
   <a href="#section-11">11</a>. References ....................................................<a href="#page-32">32</a>
      <a href="#section-11.1">11.1</a>. Normative References .....................................<a href="#page-32">32</a>
      <a href="#section-11.2">11.2</a>. Informative References ...................................<a href="#page-35">35</a>
   <a href="#appendix-A">Appendix A</a>.  Notes on Normative References ........................<a href="#page-36">36</a>









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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


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

   VoiceXML [<a href="#ref-VXML20" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.0&quot;">VXML20</a>], [<a href="#ref-VXML21" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.1&quot;">VXML21</a>] is a World Wide Web Consortium (W3C)
   standard for creating audio and video dialogs that feature
   synthesized speech, digitized audio, recognition of spoken and dual
   tone multi-frequency (DTMF) key input, recording of audio and video,
   telephony, and mixed-initiative conversations.  VoiceXML allows Web-
   based development and content delivery paradigms to be used with
   interactive video and voice response applications.

   This document describes a SIP [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] interface to VoiceXML media
   services.  Commonly, Application Servers controlling media servers
   use this protocol for pure VoiceXML processing capabilities.  SIP is
   responsible for initiating a media session to the VoiceXML media
   server and simultaneously triggering the execution of a specified
   VoiceXML application.  This protocol is an adjunct to the full
   MEDIACTRL protocol and packages mechanism.

   The interface described here leverages a mechanism for identifying
   dialog media services first described in [<a href="./rfc4240" title="&quot;Basic Network Media Services with SIP&quot;">RFC4240</a>].  The interface
   has been updated and extended to support the W3C Recommendation for
   VoiceXML 2.0 [<a href="#ref-VXML20" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.0&quot;">VXML20</a>] and VoiceXML 2.1 [<a href="#ref-VXML21" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.1&quot;">VXML21</a>].  A set of commonly
   implemented functions and extensions have been specified including
   VoiceXML dialog preparation, outbound calling, video media support,
   and transfers.  VoiceXML session variable mappings have been defined
   for SIP with an extensible mechanism for passing application-specific
   values into the VoiceXML application.  Mechanisms for returning data
   to the Application Server have also been added.

<span class="h3"><a class="selflink" id="section-1.1" href="#section-1.1">1.1</a>.  Use Cases</span>

   The VoiceXML media service user in this document is generically
   referred to as an Application Server.  In practice, it is intended
   that the interface defined by this document be applicable across a
   wide range of use cases.  Several intended use cases are described
   below.

<span class="h4"><a class="selflink" id="section-1.1.1" href="#section-1.1.1">1.1.1</a>.  IVR Services with Application Servers</span>

   SIP Application Servers provide services to users of the network.
   Typically, there may be several Application Servers in the same
   network, each specialized in providing a particular service.
   Throughout this specification and without loss of generality, we
   posit the presence of an Application Server specialized in providing
   Interactive Voice Response (IVR) services.  A typical configuration
   for this use case is illustrated below.





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                              +--------------+
                              |              |
                              |  Application |\
                              |    Server    | \
                              |              |  \ HTTP
                         SIP  +--------------+   \
                              /               \   \
             +-------------+ /             SIP \ +--------------+
             |             |/                   \|              |
             |     SIP     |                     |   VoiceXML   |
             | User Agent  |      RTP/SRTP       | Media Server |
             |             |=====================|              |
             +-------------+                     +--------------+

   Assuming the Application Server also supports HTTP, the VoiceXML
   application may be hosted on it and served up via HTTP [<a href="./rfc2616" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">RFC2616</a>].
   Note, however, that the Web model allows the VoiceXML application to
   be hosted on a separate (HTTP) Application Server from the (SIP)
   Application Server that interacts with the VoiceXML Media Server via
   this specification.  It is also possible for a static VoiceXML
   application to be stored locally on the VoiceXML Media Server,
   leveraging the VoiceXML 2.1 [<a href="#ref-VXML21" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.1&quot;">VXML21</a>] &lt;data&gt; mechanism to interact
   with a Web/Application Server when dynamic behavior is required.  The
   viability of static VoiceXML applications is further enhanced by the
   mechanisms defined in <a href="#section-2.4">Section 2.4</a>, through which the Application
   Server can make session-specific information available within the
   VoiceXML session context.

   The approach described in this document is sometimes termed the
   "delegation model" -- the Application Server is essentially
   delegating programmatic control of the human-machine interactions to
   one or more VoiceXML documents running on the VoiceXML Media Server.
   During the human-machine interactions, the Application Server remains
   in the signaling path and can respond to results returned from the
   VoiceXML Media Server or other external network events.

<span class="h4"><a class="selflink" id="section-1.1.2" href="#section-1.1.2">1.1.2</a>.  PSTN IVR Service Node</span>

   While this document is intended to enable enhanced use of VoiceXML as
   a component of larger systems and services, it is intended that
   devices that are completely unaware of this specification remain
   capable of invoking VoiceXML services offered by a VoiceXML Media
   Server compliant with this document.  A typical configuration for
   this use case is as follows:







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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


             +-------------+         SIP         +--------------+
             |             |---------------------|              |
             |   IP/PSTN   |                     |   VoiceXML   |
             |   Gateway   |      RTP/SRTP       | Media Server |
             |             |=====================|              |
             +-------------+                     +--------------+

   Note also that beyond the invocation and termination of a VoiceXML
   dialog, the semantics defined for call transfers using REFER are
   intended to be compatible with standard, existing IP/PSTN (Public
   Switched Telephone Network) gateways.

<span class="h4"><a class="selflink" id="section-1.1.3" href="#section-1.1.3">1.1.3</a>.  3GPP IMS Media Resource Function (MRF)</span>

   The 3rd Generation Partnership Project (3GPP) IP Multimedia Subsystem
   (IMS) [<a href="#ref-TS23002" title="&quot;3rd Generation Partnership Project: Network architecture (Release 6)&quot;">TS23002</a>] defines a Media Resource Function (MRF) used to offer
   media processing services such as conferencing, transcoding, and
   prompt/collect.  The capabilities offered by VoiceXML are ideal for
   offering richer media processing services in the context of the MRF.
   In this architecture, the interface defined here corresponds to the
   "Mr" interface to the MRFC (MRF Controller); the implementation of
   this interface might use separated MRFC and MRFP (MRF Processor)
   elements (as per the IMS architecture), or might be an integrated MRF
   (as is common practice).

             +----------+
             |   App    |
             |  Server  |
             +----------+
                  |
                  | SIP (ISC)
                  |
             +----------+   SIP (Mr)    +--------------+
             |  S-CSCF  |---------------|   VoiceXML   |
             |          |               |     MRF      |
             +----------+               +--------------+
                                               ||
                                               || RTP/SRTP (Mb)
                                               ||

   The above diagram is highly simplified and shows a subset of nodes
   typically involved in MRF interactions.  It should be noted that
   while the MRF will primarily be used by the Application Server via
   the Serving Call Session Control Function (S-CSCF), it is also
   possible for calls to be routed directly to the MRF without the
   involvement of an Application Server.





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   Although the above is described in terms of the 3GPP IMS
   architecture, it is intended that it is also applicable to 3GPP2,
   Next Generation Network (NGN), and PacketCable architectures that are
   converging with 3GPP IMS standards.

<span class="h4"><a class="selflink" id="section-1.1.4" href="#section-1.1.4">1.1.4</a>.  CCXML &lt;-&gt; VoiceXML Interaction</span>

   Call Control eXtensible Markup Language (CCXML) 1.0 [<a href="#ref-CCXML10" title="&quot;Voice Browser Call Control: CCXML Version 1.0&quot;">CCXML10</a>]
   applications provide services mainly through controlling the
   interaction between Connections, Conferences, and Dialogs.  Although
   CCXML is capable of supporting arbitrary dialog environments,
   VoiceXML is commonly used as a dialog environment in conjunction with
   CCXML applications; CCXML is specifically designed to effectively
   support the use of VoiceXML.  CCXML 1.0 defines language elements
   that allow for Dialogs to be prepared, started, and terminated; it
   further allows for data to be returned by the dialog environment, for
   call transfers to be requested (by the dialog) and responded to by
   the CCXML application, and for arbitrary eventing between the CCXML
   application and running dialog application.

   The interface described in this document can be used by CCXML 1.0
   implementations to control VoiceXML Media Servers.  Note, however,
   that some CCXML language features require eventing facilities between
   CCXML and VoiceXML sessions that go beyond what is defined in this
   specification.  For example, VoiceXML-controlled call transfers and
   mid-dialog, application-defined events cannot be fully realized using
   this specification alone.  A SIP event package [<a href="./rfc3265" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">RFC3265</a>] MAY be used
   in addition to this specification to provide extended eventing.

<span class="h4"><a class="selflink" id="section-1.1.5" href="#section-1.1.5">1.1.5</a>.  Other Use Cases</span>

   In addition to the use cases described in some detail above, there
   are a number of other intended use cases that are not described in
   detail, such as:

   1.  Use of a VoiceXML Media Server as an adjunct to an IP-based
       Private Branch Exchange / Automatic Call Distributor (PBX/ACD),
       possibly to provide voicemail/messaging, automated attendant, or
       other capabilities.

   2.  Invocation and control of a VoiceXML session that provides the
       voice modality component in a multimodal system.









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<span class="h3"><a class="selflink" id="section-1.2" href="#section-1.2">1.2</a>.  Terminology</span>

   Application Server:  A SIP Application Server hosts and executes
      services, in particular by terminating SIP sessions on a media
      server.  The Application Server MAY also act as an HTTP server
      [<a href="./rfc2616" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">RFC2616</a>] in interactions with media servers.

   VoiceXML Media Server:  A VoiceXML interpreter including a SIP-based
      interpreter context and the requisite media processing
      capabilities to support VoiceXML functionality.

   VoiceXML Session:  A VoiceXML Session is a multimedia session
      comprising of at least a SIP User Agent, a VoiceXML Media Server,
      the data streams between them, and an executing VoiceXML
      application.

   VoiceXML Dialog:  Equivalent to VoiceXML Session.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [<a href="./rfc2119" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">RFC2119</a>].

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  VoiceXML Session Establishment and Termination</span>

   This section describes how to establish a VoiceXML Session, with or
   without preparation, and how to terminate a session.  This section
   also addresses how session information is made available to VoiceXML
   applications.

<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>.  Service Identification</span>

   The SIP Request-URI is used to identify the VoiceXML media service.
   The user part of the SIP Request-URI is fixed to "dialog".  This is
   done to ensure compatibility with [<a href="./rfc4240" title="&quot;Basic Network Media Services with SIP&quot;">RFC4240</a>], since this document
   extends the dialog interface defined in that specification and
   because this convention from [<a href="./rfc4240" title="&quot;Basic Network Media Services with SIP&quot;">RFC4240</a>] is widely adopted by existing
   media servers.

   Standardizing the SIP Request-URI including the user part also
   improves interoperability between Application Servers and media
   servers, and reduces the provisioning overhead that would be required
   if use of a media server by an Application Server required an
   individually provisioned URI.  In this respect, this document (and
   [<a href="./rfc4240" title="&quot;Basic Network Media Services with SIP&quot;">RFC4240</a>]) do not add semantics to the user part, but rather
   standardize the way that targets on media servers are provisioned.
   Further, since Application Servers -- and not human beings -- are
   generally the clients of media servers, issues such as interpretation
   and internationalization do not apply.



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   Exposing a VoiceXML media service with a well-known address may
   enhance the possibility of exploitation: the VoiceXML Media Server is
   RECOMMENDED to use standard SIP mechanisms to authenticate endpoints
   as discussed in <a href="#section-9">Section 9</a>.

   The initial VoiceXML document is specified with the "voicexml"
   parameter.  In addition, parameters are defined that control how the
   VoiceXML Media Server fetches the specified VoiceXML document.  The
   list of parameters defined by this specification is as follows (note
   the parameter names are case-insensitive):

   voicexml:  URI of the initial VoiceXML document to fetch.  This will
      typically contain an HTTP URI, but may use other URI schemes, for
      example, to refer to local, static VoiceXML documents.  If the
      "voicexml" parameter is omitted, the VoiceXML Media Server may
      select the initial VoiceXML document by other means, such as by
      applying a default, or may reject the request.

   maxage:  Used to set the max-age value of the Cache-Control header in
      conjunction with VoiceXML documents fetched using HTTP, as per
      [<a href="./rfc2616" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">RFC2616</a>].  If omitted, the VoiceXML Media Server will use a
      default value.

   maxstale:  Used to set the max-stale value of the Cache-Control
      header in conjunction with VoiceXML documents fetched using HTTP,
      as per [<a href="./rfc2616" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">RFC2616</a>].  If omitted, the VoiceXML Media Server will use
      a default value.

   method:  Used to set the HTTP method applied in the fetch of the
      initial VoiceXML document.  Allowed values are "get" or "post"
      (case-insensitive).  Default is "get".

   postbody:  Used to set the application/x-www-form-urlencoded encoded
      [<a href="#ref-HTML4" title="&quot;HTML 4.01 Specification&quot;">HTML4</a>] HTTP body for "post" requests (or is otherwise ignored).

   ccxml:  Used to specify a "JSON value" [<a href="./rfc4627" title="&quot;The application/json Media Type for JavaScript Object Notation (JSON)&quot;">RFC4627</a>] that is mapped to
      the session.connection.ccxml VoiceXML session variable -- see
      <a href="#section-2.4">Section 2.4</a>.

   aai:  Used to specify a "JSON value" [<a href="./rfc4627" title="&quot;The application/json Media Type for JavaScript Object Notation (JSON)&quot;">RFC4627</a>] that is mapped to the
      session.connection.aai VoiceXML session variable -- see
      <a href="#section-2.4">Section 2.4</a>.

   Other application-specific parameters may be added to the Request-URI
   and are exposed in VoiceXML session variables (see <a href="#section-2.4">Section 2.4</a>).






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   Formally, the Request-URI for the VoiceXML media service has a fixed
   user part "dialog".  Seven URI parameters are defined (see the
   definition of uri-parameter in <a href="./rfc3261#section-25.1">Section&nbsp;25.1 of [RFC3261]</a>).

  dialog-param      = "voicexml=" vxml-url ; vxml-url follows the URI
                                           ; syntax defined in [<a href="./rfc3986" title="&quot;Uniform Resource Identifier (URI): Generic Syntax&quot;">RFC3986</a>]
  maxage-param      = "maxage=" 1*DIGIT
  maxstale-param    = "maxstale=" 1*DIGIT
  method-param      = "method=" ("get" / "post")
  postbody-param    = "postbody=" token
  ccxml-param       = "ccxml=" json-value
  aai-param         = "aai=" json-value
  json-value        =  false /
                       null /
                       true /
                       object /
                       array /
                       number /
                       string ; defined in [<a href="./rfc4627" title="&quot;The application/json Media Type for JavaScript Object Notation (JSON)&quot;">RFC4627</a>]

   Parameters of the Request-URI in subsequent re-INVITEs are ignored.
   One consequence of this is that the VoiceXML Media Server cannot be
   instructed by the Application Server to change the executing VoiceXML
   Application after a VoiceXML Session has been started.

   Special characters contained in the dialog-param, postbody-param,
   ccxml-param, and aai-param values must be URL-encoded ("escaped") as
   required by the SIP URI syntax, for example, '?' (%3f), '=' (%3d),
   and ';' (%3b).  The VoiceXML Media Server MUST therefore unescape
   these parameter values before making use of them or exposing them to
   running VoiceXML applications.  It is important that the VoiceXML
   Media Server only unescape the parameter values once since the
   desired VoiceXML URI value could itself be URL encoded, for example.

   Since some applications may choose to transfer confidential
   information, the VoiceXML Media Server MUST support the sips: scheme
   as discussed in <a href="#section-9">Section 9</a>.

   Informative note: With respect to the postbody-param value, since the
   application/x-www-form-urlencoded content itself escapes non-
   alphanumeric characters by inserting %HH replacements, the escaping
   rules above will result in the '%' characters being further escaped
   in addition to the '&amp;' and '=' name/value separators.








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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   As an example, the following SIP Request-URI identifies the use of
   VoiceXML media services, with
   'http://appserver.example.com/promptcollect.vxml' as the initial
   VoiceXML document, to be fetched with max-age/max-stale values of
   3600s/0s, respectively:

       sip:dialog@mediaserver.example.com; \
          voicexml=http://appserver.example.com/promptcollect.vxml; \
          maxage=3600;maxstale=0

<span class="h3"><a class="selflink" id="section-2.2" href="#section-2.2">2.2</a>.  Initiating a VoiceXML Session</span>

   A VoiceXML Session is initiated via the Application Server using a
   SIP INVITE.  Typically, the Application Server will be specialized in
   providing VoiceXML services.  At a minimum, the Application Server
   may behave as a simple proxy by rewriting the Request-URI received
   from the User Agent to a Request-URI suitable for consumption by the
   VoiceXML Media Server (as specified in <a href="#section-2.1">Section 2.1</a>).  For example, a
   User Agent might present a dialed number:

       tel:+1-201-555-0123

   that the Application Server maps to a directory assistance
   application on the VoiceXML Media Server with a Request-URI of:

       sip:dialog@ms1.example.com; \
          voicexml=http://as1.example.com/da.vxml

   Certain header values in the INVITE message to the VoiceXML Media
   Server are mapped into VoiceXML session variables and are specified
   in <a href="#section-2.4">Section 2.4</a>.

   On receipt of the INVITE, the VoiceXML Media Server issues a
   provisional response, 100 Trying, and commences the fetch of the
   initial VoiceXML document.  The 200 OK response indicates that the
   VoiceXML document has been fetched and parsed correctly and is ready
   for execution.  Application execution commences on receipt of the ACK
   (except if the dialog is being prepared as specified in <a href="#section-2.3">Section 2.3</a>).
   Note that the 100 Trying response will usually be sent on receipt of
   the INVITE in accordance with [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], since the VoiceXML Media
   Server cannot in general guarantee that the initial fetch will
   complete in less than 200 ms.  However, certain implementations may
   be able to guarantee response times to the initial INVITE, and thus
   may not need to send a 100 Trying response.

   As an optimization, prior to sending the 200 OK response, the
   VoiceXML Media Server MAY execute the application up to the point of
   the first VoiceXML waiting state or prompt flush.



<span class="grey">Burke &amp; Scott               Standards Track                    [Page 10]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-11" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   A VoiceXML Media Server, like any SIP User Agent, may be unable to
   accept the INVITE request for a variety of reasons.  For instance, a
   Session Description Protocol (SDP) offer contained in the INVITE
   might require the use of codecs that are not supported by the Media
   Server.  In such cases, the Media Server should respond as defined by
   [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].  However, there are error conditions specific to VoiceXML,
   as follows:

   1.  If the Request-URI does not conform to this specification, a 400
       Bad Request MUST be returned (unless it is used to select other
       services not defined by this specification).

   2.  If a URI parameter in the Request-URI is repeated, then the
       request MUST be rejected with a 400 Bad Request response.

   3.  If the Request-URI does not include a "voicexml" parameter, and
       the VoiceXML Media Server does not elect to use a default page,
       the VoiceXML Media Server MUST return a final response of 400 Bad
       Request, and it SHOULD include a Warning header with a 3-digit
       code of 399 and a human-readable error message.

   4.  If the VoiceXML document cannot be fetched or parsed, the
       VoiceXML Media Server MUST return a final response of 500 Server
       Internal Error and SHOULD include a Warning header with a 3-digit
       code of 399 and a human-readable error message.

   Informative note: Certain applications may pass a significant amount
   of data to the VoiceXML dialog in the form of Request-URI parameters.
   This may cause the total size of the INVITE request to exceed the MTU
   of the underlying network.  In such cases, applications/
   implementations must take care either to use a transport appropriate
   to these larger messages (such as TCP) or to use alternative means of
   passing the required information to the VoiceXML dialog (such as
   supplying a unique session identifier in the initial VoiceXML URI and
   later using that identifier as a key to retrieve data from the HTTP
   server).

<span class="h3"><a class="selflink" id="section-2.3" href="#section-2.3">2.3</a>.  Preparing a VoiceXML Session</span>

   In certain scenarios, it is beneficial to prepare a VoiceXML Session
   for execution prior to running it.  A previously prepared VoiceXML
   Session is expected to execute with minimal delay when instructed to
   do so.








<span class="grey">Burke &amp; Scott               Standards Track                    [Page 11]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-12" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   If a media-less SIP dialog is established with the initial INVITE to
   the VoiceXML Media Server, the VoiceXML application will not execute
   after receipt of the ACK.  To run the VoiceXML application, the
   Application Server (AS) must issue a re-INVITE to establish a media
   session.

   A media-less SIP dialog can be established by sending an SDP
   containing no media lines in the initial INVITE.  Alternatively, if
   no SDP is sent in the initial INVITE, the VoiceXML Media Server will
   include an offer in the 200 OK message, which can be responded to
   with an answer in the ACK with the media port(s) set to 0.

   Once a VoiceXML application is running, a re-INVITE that disables the
   media streams (i.e., sets the ports to 0) will not otherwise affect
   the executing application (except that recognition actions initiated
   while the media streams are disabled will result in noinput
   timeouts).

<span class="h3"><a class="selflink" id="section-2.4" href="#section-2.4">2.4</a>.  Session Variable Mappings</span>

   The standard VoiceXML session variables are assigned values according
   to:

   session.connection.local.uri:  Evaluates to the SIP URI specified in
      the To: header of the initial INVITE.

   session.connection.remote.uri:  Evaluates to the SIP URI specified in
      the From: header of the initial INVITE.

   session.connection.redirect:  This array is populated by information
      contained in the History-Info [<a href="./rfc4244" title="&quot;An Extension to the Session Initiation Protocol (SIP) for Request History Information&quot;">RFC4244</a>] header in the initial
      INVITE or is otherwise undefined.  Each entry (hi-entry) in the
      History-Info header is mapped, in reverse order, into an element
      of the session.connection.redirect array.  Properties of each
      element of the array are determined as follows:

      *  uri - Set to the hi-targeted-to-uri value of the History-Info
         entry

      *  pi - Set to 'true' if hi-targeted-to-uri contains a
         "Privacy=history" parameter, or if the INVITE Privacy header
         includes 'history'; 'false' otherwise

      *  si - Set to the value of the "si" parameter if it exists,
         undefined otherwise

      *  reason - Set verbatim to the value of the "Reason" parameter of
         hi-targeted-to-uri



<span class="grey">Burke &amp; Scott               Standards Track                    [Page 12]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-13" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   session.connection.protocol.name:  Evaluates to "sip".  Note that
      this is intended to reflect the use of SIP in general, and does
      not distinguish between whether the media server was accessed via
      SIP or SIPS procedures.

   session.connection.protocol.version:  Evaluates to "2.0".

   session.connection.protocol.sip.headers:  This is an associative
      array where each key in the array is the non-compact name of a SIP
      header in the initial INVITE converted to lowercase (note the case
      conversion does not apply to the header value).  If multiple
      header fields of the same field name are present, the values are
      combined into a single comma-separated value.  Implementations
      MUST at a minimum include the Call-ID header and MAY include other
      headers.  For example,
      session.connection.protocol.sip.headers["call-id"] evaluates to
      the Call-ID of the SIP dialog.

   session.connection.protocol.sip.requesturi:  This is an associative
      array where the array keys and values are formed from the URI
      parameters on the SIP Request-URI of the initial INVITE.  The
      array key is the URI parameter name converted to lowercase (note
      the case conversion does not apply to the parameter value).  The
      corresponding array value is obtained by evaluating the URI
      parameter value as a "JSON value" [<a href="./rfc4627" title="&quot;The application/json Media Type for JavaScript Object Notation (JSON)&quot;">RFC4627</a>] in the case of the
      ccxml-param and aai-param values and otherwise as a string.  In
      addition, the array's toString() function returns the full SIP
      Request-URI.  For example, assuming a Request-URI of sip:dialog@
      example.com;voicexml=http://example.com;aai=%7b"x":1%2c"y":true%7d
      then session.connection.protocol.sip.requesturi["voicexml"]
      evaluates to "http://example.com",
      session.connection.protocol.sip.requesturi["aai"].x evaluates to 1
      (type Number), session.connection.protocol.sip.requesturi["aai"].y
      evaluates to true (type Boolean), and
      session.connection.protocol.sip.requesturi evaluates to the
      complete Request-URI (type String) 'sip:dialog@
      example.com;voicexml=http://example.com;aai={"x":1,"y":true}'.

   session.connection.aai:  Evaluates to
      session.connection.protocol.sip.requesturi["aai"].

   session.connection.ccxml:  Evaluates to
      session.connection.protocol.sip.requesturi["ccxml"].








<span class="grey">Burke &amp; Scott               Standards Track                    [Page 13]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-14" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   session.connection.protocol.sip.media:  This is an array where each
      array element is an object with the following properties:

      *  type: - This required property indicates the type of the media
         associated with the stream.  The value is a string.  It is
         strongly recommended that the following values are used for
         common types of media: "audio" for audio media, and "video" for
         video media.

      *  direction: - This required property indicates the
         directionality of the media relative to
         session.connection.originator.  Defined values are sendrecv,
         sendonly, recvonly, and inactive.

      *  format: - This property is optional.  If defined, the value of
         the property is an array.  Each array element is an object that
         specifies information about one format of the media (there is
         an array element for each payload type on the m-line).  The
         object contains at least one property called "name" whose value
         is the MIME subtype of the media format (MIME subtypes are
         registered in [<a href="./rfc4855" title="&quot;Media Type Registration of RTP Payload Formats&quot;">RFC4855</a>]).  Other properties may be defined with
         string values; these correspond to required and, if defined,
         optional parameters of the format.

      As a consequence of this definition, there is an array entry in
      session.connection.protocol.sip.media for each non-disabled m-line
      for the negotiated media session.  Note that this session variable
      is updated if the media session characteristics for the VoiceXML
      Session change (i.e., due to a re-INVITE).  For an example,
      consider a connection with bidirectional G.711 mu-law "audio"
      sampled at 8 kHz.  In this case,
      session.connection.protocol.sip.media[0].type evaluates to
      "audio", session.connection.protocol.sip.media[0].direction to
      "sendrecv",
      session.connection.protocol.sip.media[0].format[0].name evaluates
      to "audio/PCMU", and
      session.connection.protocol.sip.media[0].format[0].rate evaluates
      to "8000".

   Note that when accessing SIP headers and Request-URI parameters via
   the session.connection.protocol.sip.headers and
   session.connection.protocol.sip.requesturi associative arrays defined
   above, applications can choose between two semantically equivalent
   ways of referring to the array.  For example, either of the following
   can be used to access a Request-URI parameter named "foo":

       session.connection.protocol.sip.requesturi["foo"]
       session.connection.protocol.sip.requesturi.foo



<span class="grey">Burke &amp; Scott               Standards Track                    [Page 14]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-15" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   However, it is important to note that not all SIP header names or
   Request-URI parameter names are valid ECMAScript identifiers, and as
   such, can only be accessed using the first form (array notation).
   For example, the Call-ID header can only be accessed as
   session.connection.protocol.sip.headers["call-id"]; attempting to
   access the same value as
   session.connection.protocol.sip.headers.call-id would result in an
   error.

<span class="h3"><a class="selflink" id="section-2.5" href="#section-2.5">2.5</a>.  Terminating a VoiceXML Session</span>

   The Application Server can terminate a VoiceXML Session by issuing a
   BYE to the VoiceXML Media Server.  Upon receipt of a BYE in the
   context of an existing VoiceXML Session, the VoiceXML Media Server
   MUST send a 200 OK response and MUST throw a
   'connection.disconnect.hangup' event to the VoiceXML application.  If
   the Reason header [<a href="./rfc3326" title="&quot;The Reason Header Field for the Session Initiation Protocol (SIP)&quot;">RFC3326</a>] is present on the BYE Request, then the
   value of the Reason header is provided verbatim via the '_message'
   variable within the catch element's anonymous variable scope.

   The VoiceXML Media Server may also initiate termination of the
   session by issuing a BYE request.  This will typically occur as a
   result of encountering a &lt;disconnect&gt; or &lt;exit&gt; in the VoiceXML
   application, due to the VoiceXML application running to completion,
   or due to unhandled errors within the VoiceXML application.

   See <a href="#section-4">Section 4</a> for mechanisms to return data to the Application
   Server.























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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-16" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


<span class="h3"><a class="selflink" id="section-2.6" href="#section-2.6">2.6</a>.  Examples</span>

<span class="h4"><a class="selflink" id="section-2.6.1" href="#section-2.6.1">2.6.1</a>.  Basic Session Establishment</span>

   This example illustrates an Application Server setting up a VoiceXML
   Session on behalf of a User Agent.

                         SIP               VoiceXML              HTTP
   User              Application            Media            Application
   Agent               Server               Server              Server
    |                    |                    |                    |
    |(1) INVITE [offer]  |                    |                    |
    |-------------------&gt;|(2) INVITE [offer]  |                    |
    |(3) 100 Trying      |-------------------&gt;|                    |
    |&lt;-------------------|(4) 100 Trying      |                    |
    |                    |&lt;-------------------|                    |
    |                    |                    |                    |
    |                    |                    |(5) GET             |
    |                    |                    |-------------------&gt;|
    |                    |                    |(6) 200 OK [VXML]   |
    |                    |                    |&lt;-------------------|
    |                    |                    |                    |
    |                    |(7) 200 OK [answer] |                    |
    |(8) 200 OK [answer] |&lt;-------------------|                    |
    |&lt;-------------------|                    |                    |
    |(9) ACK             |                    |                    |
    |-------------------&gt;|(10) ACK            |                    |
    |                    |-------------------&gt;| (execute           |
    |(11) RTP/SRTP       |                    |  VoiceXML          |
    |.........................................|  application)      |
    |                    |                    |                    |




















<span class="grey">Burke &amp; Scott               Standards Track                    [Page 16]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-17" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


<span class="h4"><a class="selflink" id="section-2.6.2" href="#section-2.6.2">2.6.2</a>.  VoiceXML Session Preparation</span>

   This example demonstrates the preparation of a VoiceXML Session.  In
   this example, the VoiceXML session is prepared prior to placing an
   outbound call to a User Agent, and is started as soon as the User
   Agent answers.

   The [answer1:0] notation is used to indicate an SDP answer with the
   media ports set to 0.

                         SIP               VoiceXML              HTTP
   User              Application            Media            Application
   Agent               Server               Server              Server
    |                    |                     |                    |
    |                    |(1) INVITE           |                    |
    |                    |--------------------&gt;|                    |
    |                    |(2) 100 Trying       |                    |
    |                    |&lt;--------------------|                    |
    |                    |                     |                    |
    |                    |                     |(3) GET             |
    |                    |                     |-------------------&gt;|
    |                    |                     |(4) 200 OK [VXML]   |
    |                    |                     |&lt;-------------------|
    |                    |                     |                    |
    |                    |(5) 200 OK [offer1]  |                    |
    |                    |&lt;--------------------|                    |
    |                    |(6) ACK [answer1:0]  |                    |
    |(7) INVITE          |--------------------&gt;|                    |
    |&lt;-------------------|                     |                    |
    |(8) 200 OK [offer2] |                     |                    |
    |-------------------&gt;|(9) INVITE [offer2'] |                    |
    |                    |--------------------&gt;|                    |
    |                    |(10) 100 Trying      |                    |
    |                    |&lt;--------------------|                    |
    |                    |(11) 200 OK [answer2]|                    |
    |(12) ACK [answer2]  |&lt;--------------------|                    |
    |&lt;-------------------|(13) ACK             |                    |
    |                    |--------------------&gt;| (execute           |
    |(14) RTP/SRTP                             |  VoiceXML          |
    |..........................................|  application)      |
    |                    |                     |                    |

   Implementation detail: offer2' is derived from offer2 -- it
   duplicates the m-lines and a-lines from offer2.  However, offer2'
   differs from offer2 since it must contain the same o-line as used in
   answer1:0 but with the version number incremented.  Also, if offer1
   has more m-lines than offer2, then offer2' must be padded with extra
   (rejected) m-lines.



<span class="grey">Burke &amp; Scott               Standards Track                    [Page 17]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-18" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


<span class="h4"><a class="selflink" id="section-2.6.3" href="#section-2.6.3">2.6.3</a>.  MRCP Establishment</span>

   Media Resource Control Protocol (MRCP) [<a href="#ref-MRCPv2" title="&quot;Media Resource Control Protocol Version 2 (MRCPv2)&quot;">MRCPv2</a>] is a protocol that
   enables clients such as a VoiceXML Media Server to control media
   service resources such as speech synthesizers, recognizers,
   verifiers, and identifiers residing in servers on the network.

   The example below illustrates how a VoiceXML Media Server may
   establish an MRCP session in response to an initial INVITE.

                       VoiceXML                                  HTTP
   User                Media                 MRCPv2          Application
   Agent               Server                Server             Server
    |                    |                      |                  |
    |(1) INVITE [offer1] |                      |                  |
    |-------------------&gt;|                      |                  |
    |(2) 100 Trying      |                      |                  |
    |&lt;-------------------|(3) GET               |                  |
    |                    |----------------------------------------&gt;|
    |                    |                      |                  |
    |                    |(4) 200 OK [VXML]     |                  |
    |                    |&lt;----------------------------------------|
    |                    |                      |                  |
    |                    |(5) INVITE [offer2]   |                  |
    |                    |---------------------&gt;|                  |
    |                    |                      |                  |
    |                    |(6) 200 OK [answer2]  |                  |
    |                    |&lt;---------------------|                  |
    |                    |                      |                  |
    |                    |(7) ACK               |                  |
    |                    |---------------------&gt;|                  |
    |                    |                      |                  |
    |                    |(8) MRCP connection   |                  |
    |                    |&lt;--------------------&gt;|                  |
    |(9) 200 OK [answer1]|                      |                  |
    |&lt;-------------------|                      |                  |
    |                    |                      |                  |
    |(10) ACK            |                      |                  |
    |-------------------&gt;|                      |                  |
    |                    |                      |                  |
    |(11) RTP/SRTP       |                      |                  |
    |...........................................|                  |
    |                    |                      |                  |








<span class="grey">Burke &amp; Scott               Standards Track                    [Page 18]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-19" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   In this example, the VoiceXML Media Server is responsible for
   establishing a session with the MRCPv2 Media Resource Server prior to
   sending the 200 OK response to the initial INVITE.  The VoiceXML
   Media Server will perform the appropriate offer/answer with the
   MRCPv2 Media Resource Server based on the SDP capabilities of the
   Application Server and the MRCPv2 Media Resource Server.  The
   VoiceXML Media Server will change the offer received from step 1 to
   establish an MRCPv2 session in step (5) and will re-write the SDP to
   include an m-line for each MRCPv2 resource to be used and other
   required SDP modifications as specified by MRCPv2.  Once the VoiceXML
   Media Server performs the offer/answer with the MRCPv2 Media Resource
   Server, it will establish an MRCPv2 control channel in step (8).  The
   MRCPv2 resource is deallocated when the VoiceXML Media Server
   receives or sends a BYE (not shown).

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

   This section describes the mandatory and optional media support
   required by this interface.

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

   The VoiceXML Media Server MUST support the standard offer/answer
   mechanism of [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].  In particular, if an SDP offer is not
   present in the INVITE, the VoiceXML Media Server will make an offer
   in the 200 OK response listing its supported codecs.

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

   The VoiceXML Media Server MAY support early establishment of media
   streams as described in [<a href="./rfc3960" title="&quot;Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)&quot;">RFC3960</a>].  This allows the Application
   Server to establish media streams between a User Agent and the
   VoiceXML Media Server in parallel with the initial VoiceXML document
   being processed (which may involve dynamic VoiceXML page generation
   and interaction with databases or other systems).  This is useful
   primarily for minimizing the delay in starting a VoiceXML Session,
   particularly in cases where a session with the User Agent already
   exists but the media stream associated with that session needs to be
   redirected to a VoiceXML Media Server.

   The following flow demonstrates the use of early media (using the
   Gateway model defined in [<a href="./rfc3960" title="&quot;Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)&quot;">RFC3960</a>]):









<span class="grey">Burke &amp; Scott               Standards Track                    [Page 19]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-20" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


                         SIP               VoiceXML              HTTP
   User              Application            Media            Application
   Agent               Server               Server              Server
    |                      |                   |                   |
    |..(existing session)..|                   |                   |
    |                      |(1) INVITE         |                   |
    |                      |------------------&gt;|                   |
    |                      |                   |(2) HTTP GET       |
    |                      |                   |------------------&gt;|
    |                      |(3) 183 [offer]    |                   |
    |(4) re-INVITE [offer] |&lt;------------------|                   |
    |&lt;---------------------|                   |                   |
    |(5) 200 OK [answer]   |                   |                   |
    |---------------------&gt;|                   |                   |
    |(6) ACK               |                   |                   |
    |&lt;---------------------|                   |                   |
    |                      | (7) PRACK [answer]|                   |
    |                      |------------------&gt;|                   |
    |                      | (8) PRACK 200 OK  |                   |
    |                      |&lt;------------------|                   |
    |(9) RTP/SRTP          |                   |                   |
    |..........................................|                   |
    |                      |                   |(10) 200 OK [VXML] |
    |                      |                   |&lt;------------------|
    |                      |                   |                   |
    |                      |(11) 200 OK        |                   |
    |                      |&lt;------------------|                   |
    |                      |(12) ACK           |                   |
    |                      |------------------&gt;| (execute          |
    |                      |                   |  VoiceXML         |
    |                      |                   |  application)     |
    |                      |                   |                   |

   Although [<a href="./rfc3960" title="&quot;Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)&quot;">RFC3960</a>] prefers the use of the Application Server model
   for early media over the Gateway model, the primary issue with the
   Gateway model -- forking -- is significantly less common when issuing
   requests to VoiceXML Media Servers.  This is because VoiceXML Media
   Servers respond to all requests with 200 OK responses in the absence
   of unusual errors, and they typically do so within several hundred
   milliseconds.  This makes them unlikely targets in forking scenarios,
   since alternative targets of the forking process would virtually
   never be able to respond more quickly than an automated system,
   unless they are themselves automated systems -- in which case, there
   is little point in setting up a response time race between two
   automated systems.  Issues with ringing tone generation in the
   Gateway model are also mitigated, both by the typically quick 200 OK





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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-21" ></span>
<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   response time, and because this specification mandates that no media
   packets are generated until the receipt of an ACK (thus eliminating
   the need for the User Agent to perform media packet analysis).

   Note that the offer of early media by a VoiceXML Media Server does
   not imply that the referenced VoiceXML application can always be
   fetched and executed successfully.  For instance, if the HTTP
   Application Server were to return a 4xx response in step 10 above, or
   if the provided VoiceXML content was not valid, the VoiceXML Media
   Server would still return a 500 response (as per <a href="#section-2.2">Section 2.2</a>).  At
   this point, it would be the responsibility of the Application Server
   to tear down any media streams established with the media server.

<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  Modifying the Media Session</span>

   The VoiceXML Media Server MUST allow the media session to be modified
   via a re-INVITE and SHOULD support the UPDATE method [<a href="./rfc3311" title="&quot;The Session Initiation Protocol (SIP) UPDATE Method&quot;">RFC3311</a>] for
   the same purpose.  In particular, it MUST be possible to change
   streams between sendrecv, sendonly, and recvonly as specified in
   [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>].

   Unidirectional streams are useful for announcement- or listening-only
   (hotword).  The preferred mechanism for putting the media session on
   hold is specified in [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>], i.e., the UA modifies the stream to
   be sendonly and mutes its own stream.  Modification of the media
   session does not affect VoiceXML application execution (except that
   recognition actions initiated while on hold will result in noinput
   timeouts).

<span class="h3"><a class="selflink" id="section-3.4" href="#section-3.4">3.4</a>.  Audio and Video Codecs</span>

   For the purposes of achieving a basic level of interoperability, this
   section specifies a minimal subset of codecs and RTP [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>]
   payload formats that MUST be supported by the VoiceXML Media Server.

   For audio-only applications, G.711 mu-law and A-law MUST be supported
   using the RTP payload type 0 and 8 [<a href="./rfc3551" title="&quot;RTP Profile for Audio and Video Conferences with Minimal Control&quot;">RFC3551</a>].  Other codecs and
   payload formats MAY be supported.

   Video telephony applications, which employ a video stream in addition
   to the audio stream, are possible in VoiceXML 2.0/2.1 through the use
   of multimedia file container formats such as the .3gp [<a href="#ref-TS26244" title="&quot;Transparent end-to-end packet switched streaming service (PSS); 3GPP file format (3GP)&quot;">TS26244</a>] and
   .mp4 formats [<a href="#ref-IEC14496-14" title="&quot;Information technology. Coding of audio-visual objects. MP4 file format&quot;">IEC14496-14</a>].  Video support is optional for this
   specification.  If video is supported then:

   1.  H.263 Baseline [<a href="./rfc4629" title="&quot;RTP Payload Format for ITU-T Rec&quot;">RFC4629</a>] MUST be supported.  For legacy reasons,
       the 1996 version of H.263 MAY be supported using the RTP payload
       format defined in [<a href="./rfc2190" title="&quot;RTP Payload Format for H.263 Video Streams&quot;">RFC2190</a>] (payload type 34 [<a href="./rfc3551" title="&quot;RTP Profile for Audio and Video Conferences with Minimal Control&quot;">RFC3551</a>]).



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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   2.  Adaptive Multi-Rate (AMR) narrow band audio [<a href="./rfc4867" title="&quot;RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs&quot;">RFC4867</a>] SHOULD be
       supported.

   3.  MPEG-4 video [<a href="./rfc3016" title="&quot;RTP Payload Format for MPEG-4 Audio/Visual Streams&quot;">RFC3016</a>] SHOULD be supported.

   4.  MPEG-4 Advanced Audio Coding (AAC) audio [<a href="./rfc3016" title="&quot;RTP Payload Format for MPEG-4 Audio/Visual Streams&quot;">RFC3016</a>] SHOULD be
       supported.

   5.  Other codecs and payload formats MAY be supported.

   Video record operations carried out by the VoiceXML Media Server
   typically require receipt of an intra-frame before the recording can
   commence.  The VoiceXML Media Server SHOULD use the mechanism
   described in [<a href="./rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] to request that a new intra-frame be sent.

   Since some applications may choose to transfer confidential
   information, the VoiceXML Media Server MUST support Secure RTP (SRTP)
   [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>] as discussed in <a href="#section-9">Section 9</a>.

<span class="h3"><a class="selflink" id="section-3.5" href="#section-3.5">3.5</a>.  DTMF</span>

   DTMF events [<a href="./rfc4733" title="&quot;RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals&quot;">RFC4733</a>] MUST be supported.  When the User Agent does
   not indicate support for [<a href="./rfc4733" title="&quot;RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals&quot;">RFC4733</a>], the VoiceXML Media Server MAY
   perform DTMF detection using other means such as detecting DTMF tones
   in the audio stream.  Implementation note: the reason only [<a href="./rfc4733" title="&quot;RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals&quot;">RFC4733</a>]
   telephone-events must be used when the User Agent indicates support
   of it is to avoid the risk of double detection of DTMF if detection
   on the audio stream was simultaneously applied.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Returning Data to the Application Server</span>

   This section discusses the mechanisms for returning data (e.g.,
   collected utterance or digit information) from the VoiceXML Media
   Server to the Application Server.

<span class="h3"><a class="selflink" id="section-4.1" href="#section-4.1">4.1</a>.  HTTP Mechanism</span>

   At any time during the execution of the VoiceXML application, data
   can be returned to the Application Server via HTTP using standard
   VoiceXML elements such as &lt;submit&gt; or &lt;subdialog&gt;.  Notably, the
   &lt;data&gt; element in VoiceXML 2.1 [<a href="#ref-VXML21" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.1&quot;">VXML21</a>] allows data to be sent to the
   Application Server efficiently without requiring a VoiceXML page
   transition and is ideal for short VoiceXML applications such as
   "prompt and collect".

   For most applications, it is necessary to correlate the information
   being passed over HTTP with a particular VoiceXML Session.  One way
   this can be achieved is to include the SIP Call-ID (accessible in



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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   VoiceXML via the session.connection.protocol.sip.headers array)
   within the HTTP POST fields.  Alternatively, a unique "POST-back URI"
   can be specified as an application-specific URI parameter in the
   Request-URI of the initial INVITE (accessible in VoiceXML via the
   session.connection.protocol.sip.requesturi array).

   Since some applications may choose to transfer confidential
   information, the VoiceXML Media Server MUST support the https: scheme
   as discussed in <a href="#section-9">Section 9</a>.

<span class="h3"><a class="selflink" id="section-4.2" href="#section-4.2">4.2</a>.  SIP Mechanism</span>

   Data can be returned to the Application Server via the expr or
   namelist attribute on &lt;exit&gt; or the namelist attribute on
   &lt;disconnect&gt;.  A VoiceXML Media Server MUST support encoding of the
   expr/namelist data in the message body of a BYE request sent from the
   VoiceXML Media Server as a result of encountering the &lt;exit&gt; or
   &lt;disconnect&gt; element.  A VoiceXML Media Server MAY support inclusion
   of the expr/namelist data in the message body of the 200 OK message
   in response to a received BYE request (i.e., when the VoiceXML
   application responds to the connection.disconnect.hangup event and
   subsequently executes an &lt;exit&gt; element with the expr or namelist
   attribute specified).

   Note that sending expr/namelist data in the 200 OK response requires
   that the VoiceXML Media Server delay the final response to the
   received BYE request until the VoiceXML application's post-disconnect
   final processing state terminates.  This mechanism is subject to the
   constraint that the VoiceXML Media Server must respond before the
   User Agent Client's (UAC's) timer F expires (defaults to 32 seconds).
   Moreover, for unreliable transports, the UAC will retransmit the BYE
   request according to the rules of [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].  The VoiceXML Media
   Server SHOULD implement the recommendations of [<a href="./rfc4320" title="&quot;Actions Addressing Identified Issues with the Session Initiation Protocol's (SIP) Non-INVITE Transaction&quot;">RFC4320</a>] regarding
   when to send the 100 Trying provisional response to the BYE request.

   If a VoiceXML application executes a &lt;disconnect&gt; [<a href="#ref-VXML21" title="&quot;Voice Extensible Markup Language (VoiceXML) Version 2.1&quot;">VXML21</a>] and then
   subsequently executes an &lt;exit&gt; with namelist information, the
   namelist information from the &lt;exit&gt; element is discarded.

   Namelist variables are first converted to their "JSON value"
   equivalent [<a href="./rfc4627" title="&quot;The application/json Media Type for JavaScript Object Notation (JSON)&quot;">RFC4627</a>] and encoded in the message body using the
   application/x-www-form-urlencoded format content type [<a href="#ref-HTML4" title="&quot;HTML 4.01 Specification&quot;">HTML4</a>].  The
   behavior resulting from specifying a recording variable in the
   namelist or an ECMAScript object with circular references is not
   defined.  If the expr attribute is specified on the &lt;exit&gt; element
   instead of the namelist attribute, the reserved name __exit is used.





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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   To allow the Application Server to differentiate between a BYE
   resulting from a &lt;disconnect&gt; from one resulting from an &lt;exit&gt;, the
   reserved name __reason is used, with a value of "disconnect" (without
   brackets) to reflect the use of VoiceXML's &lt;disconnect&gt; element, and
   a value of "exit" (without brackets) to an explicit &lt;exit&gt; in the
   VoiceXML document.  If the session terminates for other reasons (such
   as the media server encountering an error), this parameter may be
   omitted, or may take on platform-specific values prefixed with an
   underscore.

   This specification extends the application/x-www-form-urlencoded by
   replacing non-ASCII characters with one or more octets of the UTF-8
   representation of the character, with each octet in turn replaced by
   %HH, where HH represents the uppercase hexadecimal notation for the
   octet value and % is a literal character.  As a consequence, the
   Content-Type header field in a BYE message containing expr/namelist
   data MUST be set to application/x-www-form-urlencoded;charset=utf-8.

   The following table provides some examples of &lt;exit&gt; usage and the
   corresponding result content.

    +----------------------------------------------------------------+
    |&lt;exit&gt; Usage                  | Result Content                  |
    |------------------------------|---------------------------------|
    |&lt;exit/&gt;                       | __reason=exit                   |
    |&lt;exit expr="5"/&gt;              | __exit=5&amp;__reason=exit          |
    |&lt;exit expr="'done'"/&gt;         | __exit="done"&amp;__reason=exit     |
    |&lt;exit expr="userAuthorized"/&gt; | __exit=true&amp;__reason=exit       |
    |&lt;exit namelist="pin errors"/&gt; | pin=1234&amp;errors=0&amp;__reason=exit |
    +----------------------------------------------------------------+
    assuming the following VoiceXML variables and values:
        userAuthorized = true
        pin = 1234
        errors = 0

   For example, consider the VoiceXML snippet:

       ...
       &lt;exit namelist="id pin"/&gt;
       ...

   If id equals 1234 and pin equals 9999, say, the BYE message would
   look similar to:








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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


      BYE sip:user@pc33.example.com SIP/2.0
      Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10
      Max-Forwards: 70
      From: sip:dialog@example.com;tag=a6c85cf
      To: sip:user@example.com;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 231 BYE
      Content-Type: application/x-www-form-urlencoded;charset=utf-8
      Content-Length: 30

      id=1234&amp;pin=9999&amp;__reason=exit

   Since some applications may choose to transfer confidential
   information, the VoiceXML Media Server MUST support the S/MIME
   encoding of SIP message bodies as discussed in <a href="#section-9">Section 9</a>.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Outbound Calling</span>

   Outbound calls can be triggered via the Application Server using
   third-party call control [<a href="./rfc3725" title="&quot;Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)&quot;">RFC3725</a>].

   Flow IV from [<a href="./rfc3725" title="&quot;Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)&quot;">RFC3725</a>] is recommended in conjunction with the
   VoiceXML Session preparation mechanism.  This flow has several
   advantages over others, namely:

   1.  Selection of a VoiceXML Media Server and preparation of the
       VoiceXML application can occur before the call is placed to avoid
       the callee experiencing delays.

   2.  Avoidance of timing difficulties that could occur with other
       flows due to the time taken to fetch and parse the initial
       VoiceXML document.

   3.  The flow is IPv6 compatible.

   An example flow for an Application-Server-initiated outbound call is
   provided in <a href="#section-2.6.2">Section 2.6.2</a>.

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

   While VoiceXML is at its core a dialog language, it also provides
   optional call transfer capability.  VoiceXML's transfer capability is
   particularly suited to the PSTN IVR Service Node use case described
   in <a href="#section-1.1.2">Section 1.1.2</a>.  It is NOT RECOMMENDED to use VoiceXML's call
   transfer capability in networks involving Application Servers.
   Rather, the Application Server itself can provide call routing





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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   functionality by taking signaling actions based on the data returned
   to it from the VoiceXML Media Server via HTTP or in the SIP BYE
   message.

   If VoiceXML transfer is supported, the mechanism described in this
   section MUST be employed.  The transfer flows specified here are
   selected on the basis that they provide the best interworking across
   a wide range of SIP devices.  CCXML&lt;-&gt;VoiceXML implementations, which
   require tight-coupling in the form of bidirectional eventing to
   support all transfer types defined in VoiceXML, may benefit from
   other approaches, such as the use of SIP event packages [<a href="./rfc3265" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">RFC3265</a>].

   In what follows, the provisional responses have been omitted for
   clarity.

<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a>.  Blind</span>

   The blind-transfer sequence is initiated by the VoiceXML Media Server
   via a REFER message [<a href="./rfc3515" title="&quot;The Session Initiation Protocol (SIP) Refer Method&quot;">RFC3515</a>] on the original SIP dialog.  The
   Refer-To header contains the URI for the called party, as specified
   via the dest or destexpr attributes on the VoiceXML &lt;transfer&gt; tag.

   If the REFER request is accepted, in which case the VoiceXML Media
   Server will receive a 2xx response, the VoiceXML Media Server throws
   the connection.disconnect.transfer event and will terminate the
   VoiceXML Session with a BYE message.  For blind transfers,
   implementations MAY use [<a href="./rfc4488" title="&quot;Suppression of Session Initiation Protocol (SIP) REFER Method Implicit Subscription&quot;">RFC4488</a>] to suppress the implicit
   subscription associated with the REFER message.

   If the REFER request results in a non-2xx response, the &lt;transfer&gt;'s
   form item variable (or event raised) depends on the SIP response and
   is specified in the following table.  Note that this indicates that
   the transfer request was rejected.

    +-------------------------+-----------------------------------+
    | SIP Response            | &lt;transfer&gt; variable / event       |
    +-------------------------+-----------------------------------+
    | 404 Not Found           | error.connection.baddestination   |
    | 405 Method Not Allowed  | error.unsupported.transfer.blind  |
    | 503 Service Unavailable | error.connection.noresource       |
    | (No response)           | network_busy                      |
    | (Other 3xx/4xx/5xx/6xx) | unknown                           |
    +-------------------------+-----------------------------------+








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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   An example is illustrated below (provisional responses and NOTIFY
   messages corresponding to provisional responses have been omitted for
   clarity).

   User Agent 1        VoiceXML        User Agent 2
     (Caller)        Media Server        (Callee)
        |                 |                 |
        |(0) RTP/SRTP     |                 |
        |.................|                 |
        |                 |                 |
        |(1) REFER        | &lt;transfer&gt;      |
        |&lt;----------------|                 |
        |(2) 202 Accepted |                 |
        |----------------&gt;|                 |
        |(3) BYE          |                 |
        |&lt;----------------|                 |
        |(4) 200 OK       |                 |
        |----------------&gt;|                 |
        |                 | Stop RTP (0)    |
        |(5) INVITE                         |
        |----------------------------------&gt;|
        |(6) 200 OK                         |
        |&lt;----------------------------------|
        |(7) NOTIFY       |                 |
        |----------------&gt;|                 |
        |(8) 200 OK       |                 |
        |&lt;--------------- |                 |
        |(9) ACK                            |
        |----------------------------------&gt;|
        |(10) RTP/SRTP                      |
        |...................................|
        |                 |                 |

   If the aai or aaiexpr attribute is present on &lt;transfer&gt;, it is
   appended to the Refer-To URI as a parameter named "aai" in the REFER
   method.  Reserved characters are URL-encoded as required for SIP/SIPS
   URIs [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].  The mapping of values outside of the ASCII range is
   platform specific.

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a>.  Bridge</span>

   The bridge transfer function results in the creation of a small
   multi-party session involving the Caller, the VoiceXML Media Server,
   and the Callee.  The VoiceXML Media Server invites the Callee to the
   session and will eject the Callee if the transfer is terminated.






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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   If the aai or aaiexpr attribute is present on &lt;transfer&gt;, it is
   appended to the Request-URI in the INVITE as a URI parameter named
   "aai".  Reserved characters are URL-encoded as required for SIP/SIPS
   URIs [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].  The mapping of values outside of the ASCII range is
   platform specific.

   During the transfer attempt, audio specified in the transferaudio
   attribute of &lt;transfer&gt; is streamed to User Agent 1.  A VoiceXML
   Media Server MAY play early media received from the Callee to the
   Caller if the transferaudio attribute is omitted.

   The bridge transfer sequence is illustrated below.  The VoiceXML
   Media Server (acting as a UAC) makes a call to User Agent 2 with the
   same codecs used by User Agent 1.  When the call setup is complete,
   RTP flows between User Agent 2 and the VoiceXML Media Server.  This
   stream is mixed with User Agent 1's.

   User Agent 1         VoiceXML          User Agent 2
     (Caller)         Media Server          (Callee)
       |                   |                   |
       |(0)RTP/SRTP        |                   |
       |...................|                   |
       |                   |                   |
       |         &lt;transfer&gt;|(1)INVITE [offer]  |
       |                   |------------------&gt;|
       |                   |(2) 200 OK [answer]|
       |                   |&lt;------------------|
       |                   |(3) ACK            |
       |                   |------------------&gt;|
       |                   |(4) RTP/SRTP       |
       |              mix  |...................|
       |            (0)+(4)|                   |

   If a final response is not received from User Agent 2 from the INVITE
   and the connecttimeout expires (specified as an attribute of
   &lt;transfer&gt;), the VoiceXML Media Server will issue a CANCEL to
   terminate the transaction and the &lt;transfer&gt;'s form item variable is
   set to noanswer.

   If INVITE results in a non-2xx response, the &lt;transfer&gt;'s form item
   variable (or event raised) depends on the SIP response and is
   specified in the following table.









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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


    +-------------------------+-----------------------------------+
    | SIP Response            | &lt;transfer&gt; variable / event       |
    +-------------------------+-----------------------------------+
    | 404 Not Found           | error.connection.baddestination   |
    | 405 Method Not Allowed  | error.unsupported.transfer.bridge |
    | 408 Request Timeout     | noanswer                          |
    | 486 Busy Here           | busy                              |
    | 503 Service Unavailable | error.connection.noresource       |
    | (No response)           | network_busy                      |
    | (Other 3xx/4xx/5xx/6xx) | unknown                           |
    +-------------------------+-----------------------------------+

   Once the transfer is established, the VoiceXML Media Server can
   "listen" to the media stream from User Agent 1 to perform speech or
   DTMF hotword, which when matched results in a near-end disconnect,
   i.e., the VoiceXML Media Server issues a BYE to User Agent 2 and the
   VoiceXML application continues with User Agent 1.  A BYE will also be
   issued to User Agent 2 if the call duration exceeds the maximum
   duration specified in the maxtime attribute on &lt;transfer&gt;.

   If User Agent 2 issues a BYE during the transfer, the transfer
   terminates and the VoiceXML &lt;transfer&gt;'s form item variable receives
   the value far_end_disconnect.  If User Agent 1 issues a BYE during
   the transfer, the transfer terminates and the VoiceXML event
   connection.disconnect.transfer is thrown.

<span class="h3"><a class="selflink" id="section-6.3" href="#section-6.3">6.3</a>.  Consultation</span>

   The consultation transfer (also called attended transfer [<a href="./rfc5359" title="&quot;Session Initiation Protocol Service Examples&quot;">RFC5359</a>])
   is similar to a blind transfer except that the outcome of the
   transfer call setup is known and the Caller is not dropped as a
   result of an unsuccessful transfer attempt.

   Consultation transfer commences with the same flow as for bridge
   transfer except that the RTP streams are not mixed at step (4) and
   error.unsupported.transfer.consultation supplants
   error.unsupported.transfer.bridge.  Assuming a new SIP dialog with
   User Agent 2 is created, the remainder of the sequence follows as
   illustrated below (provisional responses and NOTIFY messages
   corresponding to provisional responses have been omitted for
   clarity).  Consultation transfer makes use of the Replaces: header
   [<a href="./rfc3891" title="&quot;The Session Initiation Protocol (SIP) &quot;">RFC3891</a>] such that User Agent 1 calls User Agent 2 and replaces the
   latter's SIP dialog with the VoiceXML Media Server with a new SIP
   dialog between the Caller and Callee.







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<span class="grey"><a href="./rfc5552">RFC 5552</a>        SIP Interface to VoiceXML Media Services        May 2009</span>


   User Agent 1        VoiceXML       User Agent 2
     (Caller)        Media Server       (Callee)
        |                 |                 |
        |(0) RTP/SRTP     |                 |
        |.................|(4) RTP/SRTP     |
        |                 |.................|
        |(5) REFER        |                 |
        |&lt;----------------|                 |
        |(6) 202 Accepted |                 |
        |----------------&gt;|                 |
        |(7) INVITE Replaces:ms1.example.com|
        |----------------------------------&gt;|
        |(8) 200 OK                         |
        |&lt;----------------------------------|
        |(9) ACK                            |
        |----------------------------------&gt;|
        |(10) RTP/SRTP                      |
        |...................................|
        |                 |(11) BYE         |
        |                 |&lt;----------------|
        |                 |(12) 200 OK      |
        |                 |----------------&gt;| Stop
        |(13) NOTIFY      |                 | RTP (4)
        |----------------&gt;|                 |
        |(14) 200 OK      |                 |
        |&lt;----------------|                 |
        |(15) BYE         |                 |
        |&lt;----------------|                 |
        |(16) 200 OK      |                 |
        |----------------&gt;| Stop            |
        |                 | RTP (0)         |

   If a response other than 202 Accepted is received in response to the
   REFER request sent to User Agent 1, the transfer terminates and an
   error.unsupported.transfer.consultation event is raised.  In
   addition, a BYE is sent to User Agent 2 to terminate the established
   outbound leg.

   The VoiceXML Media Server uses receipt of a NOTIFY message with a
   sipfrag message of 200 OK to determine that the consultation transfer
   has succeeded.  When this occurs, the connection.disconnect.transfer
   event will be thrown to the VoiceXML application, and a BYE is sent
   to User Agent 1 to terminate the session.  A NOTIFY message with a
   non-2xx final response sipfrag message body will result in the
   transfer terminating and the associated VoiceXML input item variable
   being set to 'unknown'.  Note that as a consequence of this





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   mechanism, implementations MUST NOT use [<a href="./rfc4488" title="&quot;Suppression of Session Initiation Protocol (SIP) REFER Method Implicit Subscription&quot;">RFC4488</a>] to suppress the
   implicit subscription associated with the REFER message for
   consultation transfers.

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

   The bulk of the early work for this effort was carried out on weekly
   teleconferences and over email.  The authors would particularly like
   to recognize the contributions of R. J. Auburn (Voxeo), Jeff Haynie
   (Hakano), and Scott McGlashan (Hewlett-Packard).

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

   This document owes its genesis to, "A SIP Interface to VoiceXML
   Dialog Servers", authored by J. Rosenberg, P. Mataga, and D. Ladd.
   The following people had input to the current document:

      R. J. Auburn (Voxeo)

      Hans Bjurstrom (Hewlett-Packard)

      Emily Candell (Comverse)

      Peter Danielsen (Lucent)

      Brian Frasca (Tellme)

      Jeff Haynie (Hakano)

      Scott McGlashan (Hewlett-Packard)

      Matt Oshry (Tellme)

      Rao Surapaneni (Tellme)

   The authors would like to acknowledge the support of Cullen Jennings
   and the Mediactrl chairs, Eric Burger and Spencer Dawkins.

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

   Exposing a VoiceXML media service with a well-known address may
   enhance the possibility of exploitation (for example, an invoked
   network service may trigger a billing event).  The VoiceXML Media
   Server is RECOMMENDED to use standard SIP mechanisms [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] to
   authenticate requesting endpoints and authorize per local policy.






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   Some applications may choose to transfer confidential information to
   or from the VoiceXML Media Server.  To provide data confidentiality,
   the VoiceXML Media Server MUST implement the sips: and https: schemes
   in addition to S/MIME message body encoding as described in
   [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].

   The VoiceXML Media Server MUST support Secure RTP (SRTP) [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>] to
   provide confidentiality, authentication, and replay protection for
   RTP media streams (including RTCP control traffic).

   To mitigate the possibility of denial-of-service attacks, the
   VoiceXML Media Server is RECOMMENDED (in addition to authenticating
   and authorizing endpoints described above) to provide mechanisms for
   implementing local policies such as the time-limiting of VoiceXML
   application execution.

<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a>.  IANA Considerations</span>

   IANA has registered the following parameters in the SIP/SIPS URI
   Parameters registry, following the Specification Required policy of
   [<a href="./rfc3969" title="&quot;The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)&quot;">RFC3969</a>]:

   Parameter Name    Predefined Values    Reference
   --------------    -----------------    ---------
   maxage                   No            <a href="./rfc5552">RFC 5552</a>
   maxstale                 No            <a href="./rfc5552">RFC 5552</a>
   method              "get" / "post"     <a href="./rfc5552">RFC 5552</a>
   postbody                 No            <a href="./rfc5552">RFC 5552</a>
   ccxml                    No            <a href="./rfc5552">RFC 5552</a>
   aai                      No            <a href="./rfc5552">RFC 5552</a>

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

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

   [<a id="ref-HTML4">HTML4</a>]        Raggett, D., Le Hors, A., and I. Jacobs, "HTML 4.01
                  Specification", W3C Recommendation, Dec 1999.

   [<a id="ref-RFC2119">RFC2119</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-RFC2616">RFC2616</a>]      Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
                  Masinter, L., Leach, P., and T. Berners-Lee,
                  "Hypertext Transfer Protocol -- HTTP/1.1", <a href="./rfc2616">RFC 2616</a>,
                  June 1999.






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   [<a id="ref-RFC3016">RFC3016</a>]      Kikuchi, Y., Nomura, T., Fukunaga, S., Matsui, Y., and
                  H. Kimata, "RTP Payload Format for MPEG-4 Audio/Visual
                  Streams", <a href="./rfc3016">RFC 3016</a>, November 2000.

   [<a id="ref-RFC3261">RFC3261</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-RFC3264">RFC3264</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-RFC3265">RFC3265</a>]      Roach, A., "Session Initiation Protocol (SIP)-Specific
                  Event Notification", <a href="./rfc3265">RFC 3265</a>, June 2002.

   [<a id="ref-RFC3311">RFC3311</a>]      Rosenberg, J., "The Session Initiation Protocol (SIP)
                  UPDATE Method", <a href="./rfc3311">RFC 3311</a>, October 2002.

   [<a id="ref-RFC3326">RFC3326</a>]      Schulzrinne, H., Oran, D., and G. Camarillo, "The
                  Reason Header Field for the Session Initiation
                  Protocol (SIP)", <a href="./rfc3326">RFC 3326</a>, December 2002.

   [<a id="ref-RFC3515">RFC3515</a>]      Sparks, R., "The Session Initiation Protocol (SIP)
                  Refer Method", <a href="./rfc3515">RFC 3515</a>, April 2003.

   [<a id="ref-RFC3550">RFC3550</a>]      Schulzrinne, H., Casner, S., Frederick, R., and V.
                  Jacobson, "RTP: A Transport Protocol for Real-Time
                  Applications", STD 64, <a href="./rfc3550">RFC 3550</a>, July 2003.

   [<a id="ref-RFC3551">RFC3551</a>]      Schulzrinne, H. and S. Casner, "RTP Profile for Audio
                  and Video Conferences with Minimal Control", STD 65,
                  <a href="./rfc3551">RFC 3551</a>, July 2003.

   [<a id="ref-RFC3711">RFC3711</a>]      Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
                  K. Norrman, "The Secure Real-time Transport Protocol
                  (SRTP)", <a href="./rfc3711">RFC 3711</a>, March 2004.

   [<a id="ref-RFC3725">RFC3725</a>]      Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
                  Camarillo, "Best Current Practices for Third Party
                  Call Control (3pcc) in the Session Initiation Protocol
                  (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp85">BCP 85</a>, <a href="./rfc3725">RFC 3725</a>, April 2004.

   [<a id="ref-RFC3891">RFC3891</a>]      Mahy, R., Biggs, B., and R. Dean, "The Session
                  Initiation Protocol (SIP) "Replaces" Header",
                  <a href="./rfc3891">RFC 3891</a>, September 2004.





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   [<a id="ref-RFC3986">RFC3986</a>]      Berners-Lee, T., Fielding, R., and L. Masinter,
                  "Uniform Resource Identifier (URI): Generic Syntax",
                  STD 66, <a href="./rfc3986">RFC 3986</a>, January 2005.

   [<a id="ref-RFC4244">RFC4244</a>]      Barnes, M., "An Extension to the Session Initiation
                  Protocol (SIP) for Request History Information",
                  <a href="./rfc4244">RFC 4244</a>, November 2005.

   [<a id="ref-RFC4320">RFC4320</a>]      Sparks, R., "Actions Addressing Identified Issues with
                  the Session Initiation Protocol's (SIP) Non-INVITE
                  Transaction", <a href="./rfc4320">RFC 4320</a>, January 2006.

   [<a id="ref-RFC4488">RFC4488</a>]      Levin, O., "Suppression of Session Initiation Protocol
                  (SIP) REFER Method Implicit Subscription", <a href="./rfc4488">RFC 4488</a>,
                  May 2006.

   [<a id="ref-RFC4585">RFC4585</a>]      Ott, J., Wenger, S., Sato, N., Burmeister, C., and J.
                  Rey, "Extended RTP Profile for Real-time Transport
                  Control Protocol (RTCP)-Based Feedback (RTP/AVPF)",
                  <a href="./rfc4585">RFC 4585</a>, July 2006.

   [<a id="ref-RFC4627">RFC4627</a>]      Crockford, D., "The application/json Media Type for
                  JavaScript Object Notation (JSON)", <a href="./rfc4627">RFC 4627</a>,
                  July 2006.

   [<a id="ref-RFC4629">RFC4629</a>]      Ott, H., Bormann, C., Sullivan, G., Wenger, S., and R.
                  Even, "RTP Payload Format for ITU-T Rec", <a href="./rfc4629">RFC 4629</a>,
                  January 2007.

   [<a id="ref-RFC4733">RFC4733</a>]      Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
                  Digits, Telephony Tones, and Telephony Signals",
                  <a href="./rfc4733">RFC 4733</a>, December 2006.

   [<a id="ref-RFC4855">RFC4855</a>]      Casner, S., "Media Type Registration of RTP Payload
                  Formats", <a href="./rfc4855">RFC 4855</a>, February 2007.

   [<a id="ref-RFC4867">RFC4867</a>]      Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q.
                  Xie, "RTP Payload Format and File Storage Format for
                  the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
                  Wideband (AMR-WB) Audio Codecs", <a href="./rfc4867">RFC 4867</a>, April 2007.

   [<a id="ref-VXML20">VXML20</a>]       McGlashan, S., Burnett, D., Carter, J., Danielsen, P.,
                  Ferrans, J., Hunt, A., Lucas, B., Porter, B., Rehor,
                  K., and S. Tryphonas, "Voice Extensible Markup
                  Language (VoiceXML) Version 2.0", W3C Recommendation,
                  March 2004.





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   [<a id="ref-VXML21">VXML21</a>]       Oshry, M., Auburn, R J., Baggia, P., Bodell, M.,
                  Burke, D., Burnett, D., Candell, E., Kilic, H.,
                  McGlashan, S., Lee, A., Porter, B., and K. Rehor,
                  "Voice Extensible Markup Language (VoiceXML) Version
                  2.1", W3C Candidate Recommendation, June 2005.

<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a>.  Informative References</span>

   [<a id="ref-CCXML10">CCXML10</a>]      Auburn, R J., "Voice Browser Call Control: CCXML
                  Version 1.0", W3C Working Draft, June 2005.

   [<a id="ref-IEC14496-14">IEC14496-14</a>]  "Information technology. Coding of audio-visual
                  objects. MP4 file format", ISO/IEC ISO/IEC 14496-
                  14:2003, October 2003.

   [<a id="ref-MRCPv2">MRCPv2</a>]       Shanmugham, S. and D. Burnett, "Media Resource Control
                  Protocol Version 2 (MRCPv2)", Work in Progress,
                  November 2008.

   [<a id="ref-RFC2190">RFC2190</a>]      Zhu, C., "RTP Payload Format for H.263 Video Streams",
                  <a href="./rfc2190">RFC 2190</a>, September 1997.

   [<a id="ref-RFC3960">RFC3960</a>]      Camarillo, G. and H. Schulzrinne, "Early Media and
                  Ringing Tone Generation in the Session Initiation
                  Protocol (SIP)", <a href="./rfc3960">RFC 3960</a>, December 2004.

   [<a id="ref-RFC3969">RFC3969</a>]      Camarillo, G., "The Internet Assigned Number Authority
                  (IANA) Uniform Resource Identifier (URI) Parameter
                  Registry for the Session Initiation Protocol (SIP)",
                  <a href="https://www.rfc-editor.org/bcp/bcp99">BCP 99</a>, <a href="./rfc3969">RFC 3969</a>, December 2004.

   [<a id="ref-RFC4240">RFC4240</a>]      Burger, E., Van Dyke, J., and A. Spitzer, "Basic
                  Network Media Services with SIP", <a href="./rfc4240">RFC 4240</a>,
                  December 2005.

   [<a id="ref-RFC5359">RFC5359</a>]      Johnston, A., Sparks, R., Cunningham, C., Donovan, S.,
                  and K. Summers, "Session Initiation Protocol Service
                  Examples", <a href="https://www.rfc-editor.org/bcp/bcp144">BCP 144</a>, <a href="./rfc5359">RFC 5359</a>, October 2008.

   [<a id="ref-TS23002">TS23002</a>]      "3rd Generation Partnership Project: Network
                  architecture (Release 6)", 3GPP TS 23.002 v6.6.0,
                  December 2004.

   [<a id="ref-TS26244">TS26244</a>]      "Transparent end-to-end packet switched streaming
                  service (PSS); 3GPP file format (3GP)", 3GPP TS 26.244
                  v6.4.0, December 2004.





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<span class="h2"><a class="selflink" id="appendix-A" href="#appendix-A">Appendix A</a>.  Notes on Normative References</span>

   We make a "downref" normative reference to [<a href="./rfc4627" title="&quot;The application/json Media Type for JavaScript Object Notation (JSON)&quot;">RFC4627</a>] -- an
   Informational document describing a proprietary (but extremely
   popular) format.

Authors' Addresses

   Dave Burke
   Google
   Belgrave House, 76 Buckingham Palace Road
   London  SW1W 9TQ
   United Kingdom

   EMail: daveburke@google.com


   Mark Scott
   Genesys
   1120 Finch Avenue West, 8th floor
   Toronto, Ontario  M3J 3H7
   Canada

   EMail: Mark.Scott@genesyslab.com



























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