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rfc:rfc5552

Network Working Group D. Burke Request for Comments: 5552 Google Category: Standards Track M. Scott

                                                               Genesys
                                                              May 2009
              SIP Interface to VoiceXML Media Services

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 BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents in effect on the date of
 publication of this document (http://trustee.ietf.org/license-info).
 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.

Burke & Scott Standards Track [Page 1] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

Table of Contents

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

Burke & Scott Standards Track [Page 2] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

1. Introduction

 VoiceXML [VXML20], [VXML21] 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 [RFC3261] 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 [RFC4240].  The interface
 has been updated and extended to support the W3C Recommendation for
 VoiceXML 2.0 [VXML20] and VoiceXML 2.1 [VXML21].  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.

1.1. Use Cases

 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.

1.1.1. IVR Services with Application Servers

 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.

Burke & Scott Standards Track [Page 3] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

                            +--------------+
                            |              |
                            |  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 [RFC2616].
 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 [VXML21] <data> 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 Section 2.4, 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.

1.1.2. PSTN IVR Service Node

 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:

Burke & Scott Standards Track [Page 4] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

           +-------------+         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.

1.1.3. 3GPP IMS Media Resource Function (MRF)

 The 3rd Generation Partnership Project (3GPP) IP Multimedia Subsystem
 (IMS) [TS23002] 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.

Burke & Scott Standards Track [Page 5] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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.

1.1.4. CCXML ↔ VoiceXML Interaction

 Call Control eXtensible Markup Language (CCXML) 1.0 [CCXML10]
 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 [RFC3265] MAY be used
 in addition to this specification to provide extended eventing.

1.1.5. Other Use Cases

 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.

Burke & Scott Standards Track [Page 6] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

1.2. Terminology

 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
    [RFC2616] 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 [RFC2119].

2. VoiceXML Session Establishment and Termination

 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.

2.1. Service Identification

 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 [RFC4240], since this document
 extends the dialog interface defined in that specification and
 because this convention from [RFC4240] 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
 [RFC4240]) 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.

Burke & Scott Standards Track [Page 7] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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 Section 9.
 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
    [RFC2616].  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 [RFC2616].  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
    [HTML4] HTTP body for "post" requests (or is otherwise ignored).
 ccxml:  Used to specify a "JSON value" [RFC4627] that is mapped to
    the session.connection.ccxml VoiceXML session variable -- see
    Section 2.4.
 aai:  Used to specify a "JSON value" [RFC4627] that is mapped to the
    session.connection.aai VoiceXML session variable -- see
    Section 2.4.
 Other application-specific parameters may be added to the Request-URI
 and are exposed in VoiceXML session variables (see Section 2.4).

Burke & Scott Standards Track [Page 8] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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 Section 25.1 of [RFC3261]).
dialog-param      = "voicexml=" vxml-url ; vxml-url follows the URI
                                         ; syntax defined in [RFC3986]
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 [RFC4627]
 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 Section 9.
 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 '&' and '=' name/value separators.

Burke & Scott Standards Track [Page 9] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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

2.2. Initiating a VoiceXML Session

 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 Section 2.1).  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 Section 2.4.
 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 Section 2.3).
 Note that the 100 Trying response will usually be sent on receipt of
 the INVITE in accordance with [RFC3261], 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.

Burke & Scott Standards Track [Page 10] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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
 [RFC3261].  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).

2.3. Preparing a VoiceXML Session

 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.

Burke & Scott Standards Track [Page 11] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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).

2.4. Session Variable Mappings

 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 [RFC4244] 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

Burke & Scott Standards Track [Page 12] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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" [RFC4627] 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"].

Burke & Scott Standards Track [Page 13] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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 [RFC4855]).  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

Burke & Scott Standards Track [Page 14] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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.

2.5. Terminating a VoiceXML Session

 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 [RFC3326] 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 <disconnect> or <exit> in the VoiceXML
 application, due to the VoiceXML application running to completion,
 or due to unhandled errors within the VoiceXML application.
 See Section 4 for mechanisms to return data to the Application
 Server.

Burke & Scott Standards Track [Page 15] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

2.6. Examples

2.6.1. Basic Session Establishment

 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]  |                    |                    |
  |------------------->|(2) INVITE [offer]  |                    |
  |(3) 100 Trying      |------------------->|                    |
  |<-------------------|(4) 100 Trying      |                    |
  |                    |<-------------------|                    |
  |                    |                    |                    |
  |                    |                    |(5) GET             |
  |                    |                    |------------------->|
  |                    |                    |(6) 200 OK [VXML]   |
  |                    |                    |<-------------------|
  |                    |                    |                    |
  |                    |(7) 200 OK [answer] |                    |
  |(8) 200 OK [answer] |<-------------------|                    |
  |<-------------------|                    |                    |
  |(9) ACK             |                    |                    |
  |------------------->|(10) ACK            |                    |
  |                    |------------------->| (execute           |
  |(11) RTP/SRTP       |                    |  VoiceXML          |
  |.........................................|  application)      |
  |                    |                    |                    |

Burke & Scott Standards Track [Page 16] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

2.6.2. VoiceXML Session Preparation

 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           |                    |
  |                    |-------------------->|                    |
  |                    |(2) 100 Trying       |                    |
  |                    |<--------------------|                    |
  |                    |                     |                    |
  |                    |                     |(3) GET             |
  |                    |                     |------------------->|
  |                    |                     |(4) 200 OK [VXML]   |
  |                    |                     |<-------------------|
  |                    |                     |                    |
  |                    |(5) 200 OK [offer1]  |                    |
  |                    |<--------------------|                    |
  |                    |(6) ACK [answer1:0]  |                    |
  |(7) INVITE          |-------------------->|                    |
  |<-------------------|                     |                    |
  |(8) 200 OK [offer2] |                     |                    |
  |------------------->|(9) INVITE [offer2'] |                    |
  |                    |-------------------->|                    |
  |                    |(10) 100 Trying      |                    |
  |                    |<--------------------|                    |
  |                    |(11) 200 OK [answer2]|                    |
  |(12) ACK [answer2]  |<--------------------|                    |
  |<-------------------|(13) ACK             |                    |
  |                    |-------------------->| (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.

Burke & Scott Standards Track [Page 17] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

2.6.3. MRCP Establishment

 Media Resource Control Protocol (MRCP) [MRCPv2] 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] |                      |                  |
  |------------------->|                      |                  |
  |(2) 100 Trying      |                      |                  |
  |<-------------------|(3) GET               |                  |
  |                    |---------------------------------------->|
  |                    |                      |                  |
  |                    |(4) 200 OK [VXML]     |                  |
  |                    |<----------------------------------------|
  |                    |                      |                  |
  |                    |(5) INVITE [offer2]   |                  |
  |                    |--------------------->|                  |
  |                    |                      |                  |
  |                    |(6) 200 OK [answer2]  |                  |
  |                    |<---------------------|                  |
  |                    |                      |                  |
  |                    |(7) ACK               |                  |
  |                    |--------------------->|                  |
  |                    |                      |                  |
  |                    |(8) MRCP connection   |                  |
  |                    |<-------------------->|                  |
  |(9) 200 OK [answer1]|                      |                  |
  |<-------------------|                      |                  |
  |                    |                      |                  |
  |(10) ACK            |                      |                  |
  |------------------->|                      |                  |
  |                    |                      |                  |
  |(11) RTP/SRTP       |                      |                  |
  |...........................................|                  |
  |                    |                      |                  |

Burke & Scott Standards Track [Page 18] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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).

3. Media Support

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

3.1. Offer/Answer

 The VoiceXML Media Server MUST support the standard offer/answer
 mechanism of [RFC3264].  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.

3.2. Early Media

 The VoiceXML Media Server MAY support early establishment of media
 streams as described in [RFC3960].  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 [RFC3960]):

Burke & Scott Standards Track [Page 19] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

                       SIP               VoiceXML              HTTP
 User              Application            Media            Application
 Agent               Server               Server              Server
  |                      |                   |                   |
  |..(existing session)..|                   |                   |
  |                      |(1) INVITE         |                   |
  |                      |------------------>|                   |
  |                      |                   |(2) HTTP GET       |
  |                      |                   |------------------>|
  |                      |(3) 183 [offer]    |                   |
  |(4) re-INVITE [offer] |<------------------|                   |
  |<---------------------|                   |                   |
  |(5) 200 OK [answer]   |                   |                   |
  |--------------------->|                   |                   |
  |(6) ACK               |                   |                   |
  |<---------------------|                   |                   |
  |                      | (7) PRACK [answer]|                   |
  |                      |------------------>|                   |
  |                      | (8) PRACK 200 OK  |                   |
  |                      |<------------------|                   |
  |(9) RTP/SRTP          |                   |                   |
  |..........................................|                   |
  |                      |                   |(10) 200 OK [VXML] |
  |                      |                   |<------------------|
  |                      |                   |                   |
  |                      |(11) 200 OK        |                   |
  |                      |<------------------|                   |
  |                      |(12) ACK           |                   |
  |                      |------------------>| (execute          |
  |                      |                   |  VoiceXML         |
  |                      |                   |  application)     |
  |                      |                   |                   |
 Although [RFC3960] 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

Burke & Scott Standards Track [Page 20] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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 Section 2.2).  At
 this point, it would be the responsibility of the Application Server
 to tear down any media streams established with the media server.

3.3. Modifying the Media Session

 The VoiceXML Media Server MUST allow the media session to be modified
 via a re-INVITE and SHOULD support the UPDATE method [RFC3311] for
 the same purpose.  In particular, it MUST be possible to change
 streams between sendrecv, sendonly, and recvonly as specified in
 [RFC3264].
 Unidirectional streams are useful for announcement- or listening-only
 (hotword).  The preferred mechanism for putting the media session on
 hold is specified in [RFC3264], 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).

3.4. Audio and Video Codecs

 For the purposes of achieving a basic level of interoperability, this
 section specifies a minimal subset of codecs and RTP [RFC3550]
 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 [RFC3551].  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 [TS26244] and
 .mp4 formats [IEC14496-14].  Video support is optional for this
 specification.  If video is supported then:
 1.  H.263 Baseline [RFC4629] MUST be supported.  For legacy reasons,
     the 1996 version of H.263 MAY be supported using the RTP payload
     format defined in [RFC2190] (payload type 34 [RFC3551]).

Burke & Scott Standards Track [Page 21] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 2.  Adaptive Multi-Rate (AMR) narrow band audio [RFC4867] SHOULD be
     supported.
 3.  MPEG-4 video [RFC3016] SHOULD be supported.
 4.  MPEG-4 Advanced Audio Coding (AAC) audio [RFC3016] 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 [RFC4585] 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)
 [RFC3711] as discussed in Section 9.

3.5. DTMF

 DTMF events [RFC4733] MUST be supported.  When the User Agent does
 not indicate support for [RFC4733], 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 [RFC4733]
 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.

4. Returning Data to the Application Server

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

4.1. HTTP Mechanism

 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 <submit> or <subdialog>.  Notably, the
 <data> element in VoiceXML 2.1 [VXML21] 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

Burke & Scott Standards Track [Page 22] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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 Section 9.

4.2. SIP Mechanism

 Data can be returned to the Application Server via the expr or
 namelist attribute on <exit> or the namelist attribute on
 <disconnect>.  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 <exit> or
 <disconnect> 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 <exit> 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 [RFC3261].  The VoiceXML Media
 Server SHOULD implement the recommendations of [RFC4320] regarding
 when to send the 100 Trying provisional response to the BYE request.
 If a VoiceXML application executes a <disconnect> [VXML21] and then
 subsequently executes an <exit> with namelist information, the
 namelist information from the <exit> element is discarded.
 Namelist variables are first converted to their "JSON value"
 equivalent [RFC4627] and encoded in the message body using the
 application/x-www-form-urlencoded format content type [HTML4].  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 <exit> element
 instead of the namelist attribute, the reserved name __exit is used.

Burke & Scott Standards Track [Page 23] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 To allow the Application Server to differentiate between a BYE
 resulting from a <disconnect> from one resulting from an <exit>, the
 reserved name __reason is used, with a value of "disconnect" (without
 brackets) to reflect the use of VoiceXML's <disconnect> element, and
 a value of "exit" (without brackets) to an explicit <exit> 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 <exit> usage and the
 corresponding result content.
  +----------------------------------------------------------------+
  |<exit> Usage                  | Result Content                  |
  |------------------------------|---------------------------------|
  |<exit/>                       | __reason=exit                   |
  |<exit expr="5"/>              | __exit=5&__reason=exit          |
  |<exit expr="'done'"/>         | __exit="done"&__reason=exit     |
  |<exit expr="userAuthorized"/> | __exit=true&__reason=exit       |
  |<exit namelist="pin errors"/> | pin=1234&errors=0&__reason=exit |
  +----------------------------------------------------------------+
  assuming the following VoiceXML variables and values:
      userAuthorized = true
      pin = 1234
      errors = 0
 For example, consider the VoiceXML snippet:
     ...
     <exit namelist="id pin"/>
     ...
 If id equals 1234 and pin equals 9999, say, the BYE message would
 look similar to:

Burke & Scott Standards Track [Page 24] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

    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&pin=9999&__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 Section 9.

5. Outbound Calling

 Outbound calls can be triggered via the Application Server using
 third-party call control [RFC3725].
 Flow IV from [RFC3725] 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 Section 2.6.2.

6. Call Transfer

 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 Section 1.1.2.  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

Burke & Scott Standards Track [Page 25] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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<->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 [RFC3265].
 In what follows, the provisional responses have been omitted for
 clarity.

6.1. Blind

 The blind-transfer sequence is initiated by the VoiceXML Media Server
 via a REFER message [RFC3515] 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 <transfer> 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 [RFC4488] to suppress the implicit
 subscription associated with the REFER message.
 If the REFER request results in a non-2xx response, the <transfer>'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            | <transfer> 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                           |
  +-------------------------+-----------------------------------+

Burke & Scott Standards Track [Page 26] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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        | <transfer>      |
      |<----------------|                 |
      |(2) 202 Accepted |                 |
      |---------------->|                 |
      |(3) BYE          |                 |
      |<----------------|                 |
      |(4) 200 OK       |                 |
      |---------------->|                 |
      |                 | Stop RTP (0)    |
      |(5) INVITE                         |
      |---------------------------------->|
      |(6) 200 OK                         |
      |<----------------------------------|
      |(7) NOTIFY       |                 |
      |---------------->|                 |
      |(8) 200 OK       |                 |
      |<--------------- |                 |
      |(9) ACK                            |
      |---------------------------------->|
      |(10) RTP/SRTP                      |
      |...................................|
      |                 |                 |
 If the aai or aaiexpr attribute is present on <transfer>, 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 [RFC3261].  The mapping of values outside of the ASCII range is
 platform specific.

6.2. Bridge

 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.

Burke & Scott Standards Track [Page 27] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 If the aai or aaiexpr attribute is present on <transfer>, 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 [RFC3261].  The mapping of values outside of the ASCII range is
 platform specific.
 During the transfer attempt, audio specified in the transferaudio
 attribute of <transfer> 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        |                   |
     |...................|                   |
     |                   |                   |
     |         <transfer>|(1)INVITE [offer]  |
     |                   |------------------>|
     |                   |(2) 200 OK [answer]|
     |                   |<------------------|
     |                   |(3) ACK            |
     |                   |------------------>|
     |                   |(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
 <transfer>), the VoiceXML Media Server will issue a CANCEL to
 terminate the transaction and the <transfer>'s form item variable is
 set to noanswer.
 If INVITE results in a non-2xx response, the <transfer>'s form item
 variable (or event raised) depends on the SIP response and is
 specified in the following table.

Burke & Scott Standards Track [Page 28] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

  +-------------------------+-----------------------------------+
  | SIP Response            | <transfer> 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 <transfer>.
 If User Agent 2 issues a BYE during the transfer, the transfer
 terminates and the VoiceXML <transfer>'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.

6.3. Consultation

 The consultation transfer (also called attended transfer [RFC5359])
 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
 [RFC3891] 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.

Burke & Scott Standards Track [Page 29] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

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

Burke & Scott Standards Track [Page 30] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 mechanism, implementations MUST NOT use [RFC4488] to suppress the
 implicit subscription associated with the REFER message for
 consultation transfers.

7. Contributors

 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).

8. Acknowledgements

 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.

9. Security Considerations

 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 [RFC3261] to
 authenticate requesting endpoints and authorize per local policy.

Burke & Scott Standards Track [Page 31] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 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
 [RFC3261].
 The VoiceXML Media Server MUST support Secure RTP (SRTP) [RFC3711] 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.

10. IANA Considerations

 IANA has registered the following parameters in the SIP/SIPS URI
 Parameters registry, following the Specification Required policy of
 [RFC3969]:
 Parameter Name    Predefined Values    Reference
 --------------    -----------------    ---------
 maxage                   No            RFC 5552
 maxstale                 No            RFC 5552
 method              "get" / "post"     RFC 5552
 postbody                 No            RFC 5552
 ccxml                    No            RFC 5552
 aai                      No            RFC 5552

11. References

11.1. Normative References

 [HTML4]        Raggett, D., Le Hors, A., and I. Jacobs, "HTML 4.01
                Specification", W3C Recommendation, Dec 1999.
 [RFC2119]      Bradner, S., "Key words for use in RFCs to Indicate
                Requirement Levels", BCP 14, RFC 2119, March 1997.
 [RFC2616]      Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
                Masinter, L., Leach, P., and T. Berners-Lee,
                "Hypertext Transfer Protocol -- HTTP/1.1", RFC 2616,
                June 1999.

Burke & Scott Standards Track [Page 32] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 [RFC3016]      Kikuchi, Y., Nomura, T., Fukunaga, S., Matsui, Y., and
                H. Kimata, "RTP Payload Format for MPEG-4 Audio/Visual
                Streams", RFC 3016, November 2000.
 [RFC3261]      Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                Johnston, A., Peterson, J., Sparks, R., Handley, M.,
                and E. Schooler, "SIP: Session Initiation Protocol",
                RFC 3261, June 2002.
 [RFC3264]      Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
                Model with Session Description Protocol (SDP)",
                RFC 3264, June 2002.
 [RFC3265]      Roach, A., "Session Initiation Protocol (SIP)-Specific
                Event Notification", RFC 3265, June 2002.
 [RFC3311]      Rosenberg, J., "The Session Initiation Protocol (SIP)
                UPDATE Method", RFC 3311, October 2002.
 [RFC3326]      Schulzrinne, H., Oran, D., and G. Camarillo, "The
                Reason Header Field for the Session Initiation
                Protocol (SIP)", RFC 3326, December 2002.
 [RFC3515]      Sparks, R., "The Session Initiation Protocol (SIP)
                Refer Method", RFC 3515, April 2003.
 [RFC3550]      Schulzrinne, H., Casner, S., Frederick, R., and V.
                Jacobson, "RTP: A Transport Protocol for Real-Time
                Applications", STD 64, RFC 3550, July 2003.
 [RFC3551]      Schulzrinne, H. and S. Casner, "RTP Profile for Audio
                and Video Conferences with Minimal Control", STD 65,
                RFC 3551, July 2003.
 [RFC3711]      Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
                K. Norrman, "The Secure Real-time Transport Protocol
                (SRTP)", RFC 3711, March 2004.
 [RFC3725]      Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
                Camarillo, "Best Current Practices for Third Party
                Call Control (3pcc) in the Session Initiation Protocol
                (SIP)", BCP 85, RFC 3725, April 2004.
 [RFC3891]      Mahy, R., Biggs, B., and R. Dean, "The Session
                Initiation Protocol (SIP) "Replaces" Header",
                RFC 3891, September 2004.

Burke & Scott Standards Track [Page 33] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 [RFC3986]      Berners-Lee, T., Fielding, R., and L. Masinter,
                "Uniform Resource Identifier (URI): Generic Syntax",
                STD 66, RFC 3986, January 2005.
 [RFC4244]      Barnes, M., "An Extension to the Session Initiation
                Protocol (SIP) for Request History Information",
                RFC 4244, November 2005.
 [RFC4320]      Sparks, R., "Actions Addressing Identified Issues with
                the Session Initiation Protocol's (SIP) Non-INVITE
                Transaction", RFC 4320, January 2006.
 [RFC4488]      Levin, O., "Suppression of Session Initiation Protocol
                (SIP) REFER Method Implicit Subscription", RFC 4488,
                May 2006.
 [RFC4585]      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)",
                RFC 4585, July 2006.
 [RFC4627]      Crockford, D., "The application/json Media Type for
                JavaScript Object Notation (JSON)", RFC 4627,
                July 2006.
 [RFC4629]      Ott, H., Bormann, C., Sullivan, G., Wenger, S., and R.
                Even, "RTP Payload Format for ITU-T Rec", RFC 4629,
                January 2007.
 [RFC4733]      Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
                Digits, Telephony Tones, and Telephony Signals",
                RFC 4733, December 2006.
 [RFC4855]      Casner, S., "Media Type Registration of RTP Payload
                Formats", RFC 4855, February 2007.
 [RFC4867]      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", RFC 4867, April 2007.
 [VXML20]       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.

Burke & Scott Standards Track [Page 34] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

 [VXML21]       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.

11.2. Informative References

 [CCXML10]      Auburn, R J., "Voice Browser Call Control: CCXML
                Version 1.0", W3C Working Draft, June 2005.
 [IEC14496-14]  "Information technology. Coding of audio-visual
                objects. MP4 file format", ISO/IEC ISO/IEC 14496-
                14:2003, October 2003.
 [MRCPv2]       Shanmugham, S. and D. Burnett, "Media Resource Control
                Protocol Version 2 (MRCPv2)", Work in Progress,
                November 2008.
 [RFC2190]      Zhu, C., "RTP Payload Format for H.263 Video Streams",
                RFC 2190, September 1997.
 [RFC3960]      Camarillo, G. and H. Schulzrinne, "Early Media and
                Ringing Tone Generation in the Session Initiation
                Protocol (SIP)", RFC 3960, December 2004.
 [RFC3969]      Camarillo, G., "The Internet Assigned Number Authority
                (IANA) Uniform Resource Identifier (URI) Parameter
                Registry for the Session Initiation Protocol (SIP)",
                BCP 99, RFC 3969, December 2004.
 [RFC4240]      Burger, E., Van Dyke, J., and A. Spitzer, "Basic
                Network Media Services with SIP", RFC 4240,
                December 2005.
 [RFC5359]      Johnston, A., Sparks, R., Cunningham, C., Donovan, S.,
                and K. Summers, "Session Initiation Protocol Service
                Examples", BCP 144, RFC 5359, October 2008.
 [TS23002]      "3rd Generation Partnership Project: Network
                architecture (Release 6)", 3GPP TS 23.002 v6.6.0,
                December 2004.
 [TS26244]      "Transparent end-to-end packet switched streaming
                service (PSS); 3GPP file format (3GP)", 3GPP TS 26.244
                v6.4.0, December 2004.

Burke & Scott Standards Track [Page 35] RFC 5552 SIP Interface to VoiceXML Media Services May 2009

Appendix A. Notes on Normative References

 We make a "downref" normative reference to [RFC4627] -- 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

Burke & Scott Standards Track [Page 36]

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