[Docs] [txt|pdf] [Tracker] [Email] [Diff1] [Diff2] [Nits]

Versions: 00 01 02 03

Network Working Group                                           D. Burke
Internet-Draft                                                    Google
Intended status: Informational                                  M. Scott
Expires: January 10, 2008                                        Genesys
                                                               J. Haynie
                                                              Hakano Inc
                                                               R. Auburn
                                                                   Voxeo
                                                            S. McGlashan
                                                         Hewlett-Packard
                                                            July 9, 2007


                SIP Interface to VoiceXML Media Services
                        draft-burke-vxml-03.txt

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on January 10, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2007).







Burke, et al.           Expires January 10, 2008                [Page 1]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


Abstract

   This document describes a SIP interface to VoiceXML media services,
   which is commonly employed between application servers and media
   servers offering VoiceXML processing capabilities.














































Burke, et al.           Expires January 10, 2008                [Page 2]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


Comments

   Comments are solicited and should be addressed to the authors.
















































Burke, et al.           Expires January 10, 2008                [Page 3]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  5
     1.1.  Use Cases  . . . . . . . . . . . . . . . . . . . . . . . .  5
       1.1.1.  IVR Services with Application Servers  . . . . . . . .  5
       1.1.2.  PSTN IVR Service Node  . . . . . . . . . . . . . . . .  6
       1.1.3.  3GPP IMS Media Resource Function (MRF) . . . . . . . .  7
       1.1.4.  CCXML <-> VoiceXML Interaction . . . . . . . . . . . .  8
       1.1.5.  Other Use Cases  . . . . . . . . . . . . . . . . . . .  8
     1.2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . .  8
   2.  VoiceXML Session Establishment and Termination . . . . . . . . 10
     2.1.  Service Identification . . . . . . . . . . . . . . . . . . 10
     2.2.  Initiating a VoiceXML Session  . . . . . . . . . . . . . . 12
     2.3.  Preparing a VoiceXML Session . . . . . . . . . . . . . . . 14
     2.4.  Session Variable Mappings  . . . . . . . . . . . . . . . . 14
     2.5.  Terminating a VoiceXML Session . . . . . . . . . . . . . . 17
     2.6.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . 18
       2.6.1.  Basic Session Establishment  . . . . . . . . . . . . . 18
       2.6.2.  VoiceXML Session Preparation . . . . . . . . . . . . . 18
       2.6.3.  MRCP Establishment . . . . . . . . . . . . . . . . . . 19
   3.  Media Support  . . . . . . . . . . . . . . . . . . . . . . . . 22
     3.1.  Offer/Answer . . . . . . . . . . . . . . . . . . . . . . . 22
     3.2.  Early Media  . . . . . . . . . . . . . . . . . . . . . . . 22
     3.3.  Modifying the Media Session  . . . . . . . . . . . . . . . 24
     3.4.  Audio and Video Codecs . . . . . . . . . . . . . . . . . . 24
     3.5.  DTMF . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
   4.  Returning Data to the Application Server . . . . . . . . . . . 26
     4.1.  HTTP Mechanism . . . . . . . . . . . . . . . . . . . . . . 26
     4.2.  SIP Mechanism  . . . . . . . . . . . . . . . . . . . . . . 26
   5.  Outbound Calling . . . . . . . . . . . . . . . . . . . . . . . 29
     5.1.  Third Party Call Control Mechanism . . . . . . . . . . . . 29
     5.2.  REFER Mechanism  . . . . . . . . . . . . . . . . . . . . . 29
   6.  Call Transfer  . . . . . . . . . . . . . . . . . . . . . . . . 31
     6.1.  Blind  . . . . . . . . . . . . . . . . . . . . . . . . . . 31
     6.2.  Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . 33
     6.3.  Consultation . . . . . . . . . . . . . . . . . . . . . . . 34
   7.  Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 37
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 38
   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 39
   10. Changes since last version:  . . . . . . . . . . . . . . . . . 40
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 41
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 41
     11.2. Informative References . . . . . . . . . . . . . . . . . . 43
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 45
   Intellectual Property and Copyright Statements . . . . . . . . . . 46






Burke, et al.           Expires January 10, 2008                [Page 4]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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 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, which is commonly employed between Application Servers and
   media servers offering 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.

   The interface described here owes its genesis to the draft [SIPVXML]
   and leverages a mechanism for identifying dialog media services
   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 is 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 specialised in providing a particular service.
   Throughout this specification and without loss of generality, we
   posit the presence of an Application Server specialised in providing
   IVR services.  A typical configuration for this use case is
   illustrated below.





Burke, et al.           Expires January 10, 2008                [Page 5]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


                              +--------------+
                              |              |
                              |  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 but that
   support [RFC4240] 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, et al.           Expires January 10, 2008                [Page 6]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


             +-------------+         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 gateways.

1.1.3.  3GPP IMS Media Resource Function (MRF)

   The 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;
   the implementation of this interface might use separated MRFC and
   MRFP 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 S-CSCF, it is also possible for calls to be routed directly to
   the MRF without the involvement of an Application Server.

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



Burke, et al.           Expires January 10, 2008                [Page 7]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


1.1.4.  CCXML <-> VoiceXML Interaction

   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 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.

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.





Burke, et al.           Expires January 10, 2008                [Page 8]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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].









































Burke, et al.           Expires January 10, 2008                [Page 9]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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 as
   defined in [RFC4240].  The user part of the SIP Request-URI is fixed
   to "dialog".  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:

   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).
      The postbody value is the prepared application/
      x-www-form-urlencoded content, subsequently URL-encoded (see note
      below).

   Other application-specific parameters may be added to the Request-URI
   and are exposed in VoiceXML session variables (see section 2.4).

   The BNF for the Request-URI is given below:



Burke, et al.           Expires January 10, 2008               [Page 10]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


    DIALOG-URL        = sip-ind dialog-ind "@" hostport
                        dialog-parameters

    sip-ind           = "sip:" / "sips:"
    dialog-ind        = "dialog"

    dialog-parameters = [ init-parameters ]
                        [ vxml-parameters ]
                        [ uri-parameters ]

    init-parameters   = init-param [ init-parameters ]

    init-param        = ";" (dialog-param /
                             maxage-param /
                             maxstale-param /
                             method-param /
                             postbody-param)

    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

    vxml-parameters   = vxml-param [ vxml-parameters ]

    vxml-param        = ";" vxml-keyword "=" vxml-value

    vxml-keyword      = token

    vxml-value        =  false /
                         null /
                         true /
                         object /
                         array /
                         number /
                         string ; see 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.




Burke, et al.           Expires January 10, 2008               [Page 11]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   Incorrectly formed requests MUST be rejected with the appropriate 4xx
   class response.  If one of the init-parameters is repeated, then the
   request MUST be rejected with a 400 Bad Request response.

   Note: Special characters in Request-URI parameter values need to be
   URL-encoded as required by the SIP URI syntax, for example '?' (%3f),
   '=' (%3d), and ';' (%3b).  The VoiceXML Media Server MUST therefore
   unescape Request-URI 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.  When a postbody is included, its entire content including
   any line breaks (represented by a CR LF pair) is encoded as a single
   parameter value following the above rules (such that the line breaks
   would be replaced by '%0D%0A', for example).

   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 or REFER (see section 5.2).  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

   which 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

   The Application Server SHOULD insert its own URI in the Record-Route
   header so that it remains in the signaling path for subsequent
   signaling related to the session.  This is of particular importance



Burke, et al.           Expires January 10, 2008               [Page 12]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   for call transfers so that upstream Application Servers or proxy
   servers see signaling originating from the Application Server and not
   the VoiceXML Media Server itself.  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.

   A VoiceXML Media Server, like any SIP User Agent, may be unable to
   accept the INVITE request for a variety of reasons.  For instance, an
   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 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 SHOULD include a Warning header with a 3-digit code
       of 399 and a human readable error message.

   3.  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.

   Informational 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



Burke, et al.           Expires January 10, 2008               [Page 13]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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).  This note also applies if the dialog is
   started using a REFER request as described in section 5.2.

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.

   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 AS
   must issue a re-INVITE to establish a media session.

   A media-less SIP dialog can be established by sending 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 which 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 (or REFER).

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

   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



Burke, et al.           Expires January 10, 2008               [Page 14]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


      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

   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 lower-case (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 (or
      REFER).  The array key is the URI parameter name.  The
      corresponding array value is derived from the URI parameter value
      according to the following rules:

      *  If the URI parameter name is an init-param or dialog-param, the
         corresponding array value is obtained by evaluating the URI
         parameter value as a string.

      *  If the URI parameter name is a vxml-param, the corresponding
         array value is obtained by evaluating the URI parameter value
         as a "JSON value" [RFC4627].



Burke, et al.           Expires January 10, 2008               [Page 15]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


      *  If the URI parameter name is present but its value is omitted,
         the value is an empty 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://ajax.com;obj={"x":1,"y":true} then
      session.connection.protocol.sip.requesturi["voicexml"] evaluates
      to "http://ajax.com",
      session.connection.protocol.sip.requesturi["obj"].x evaluates to 1
      (type Number), session.connection.protocol.sip.requesturi["obj"].y
      evaluates to true (type Boolean), and
      session.connection.protocol.sip.requesturi evaluates to the
      complete Request-URI.

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

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

   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
         which 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



Burke, et al.           Expires January 10, 2008               [Page 16]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


      Session change (i.e. due to a re-INVITE).  For an example,
      consider a connection with bi-directional G.711 mu-law audio
      sampled at 8kHz.  In this case,
      session.connection.protocol.sip.media[0].type evaluates to
      "audio", session.connection.protocol.sip.media[0].direction to
      "sendrecv", and
      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

   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 encoutering 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



Burke, et al.           Expires January 10, 2008               [Page 17]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   Server.

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)      |
    |                    |                    |                    |


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.






Burke, et al.           Expires January 10, 2008               [Page 18]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


                         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)      |
    |                    |                     |                    |


2.6.3.  MRCP Establishment

   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.









Burke, et al.           Expires January 10, 2008               [Page 19]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


                       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       |                      |                  |
   .............................................|                  |
    |                    |                      |                  |


   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 a 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 a MRCPv2 control channel in step (8).  The
   MRCPv2 resource is deallocated when the VoiceXML Media Server



Burke, et al.           Expires January 10, 2008               [Page 20]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   receives or sends a BYE (not shown).


















































Burke, et al.           Expires January 10, 2008               [Page 21]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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 by sending a 183 Session Progress provisional response to the
   initial INVITE.  This allows the Application Server to establish
   media streams between a user agent and the VoiceXML Media Server
   while the initial VoiceXML document is being processed.  This is
   useful primarily for minimizing the delay in starting a VoiceXML
   Session, since media stream establishment and initial VoiceXML
   document processing can occur in parallel.  This can be particularly
   important in cases where the session with the user agent has already
   been established, since the user agent is already "connected".  The
   following flow demonstrates the use of early media:


























Burke, et al.           Expires January 10, 2008               [Page 22]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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


   In the figure shown above, although step 9 (HTTP GET) is shown
   occuring after the early media offer/answer exchange (starting in
   step 2), the intent is that the fetching of the VoiceXML document
   happens concurrently with the negotiation of early media.

   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 VoiceML 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.




Burke, et al.           Expires January 10, 2008               [Page 23]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   The use of early media is substantially complicated if the SDP
   supplied in the 183 Session Progress differs from that supplied in
   the 200 OK.  Therefore, if a VoiceXML Media Server generates a 183
   Session Progress provisional response containing SDP, it MUST return
   identical SDP when generating the 200 OK final response (i.e. the
   "gateway model" in [RFC3960]).

   Early media is not optimal in all circumstances; for instance, when
   handling an incoming call, a 183 Session Progress propagated by the
   Application Server to the user agent will typically stop the
   "ringback tone" a user would otherwise hear.  Furthermore, a 183
   Session Progress provisional response does not guarantee that the
   VoiceXML application will be executed successfully - the subsequent
   fetching of the VoiceXML document could fail.

   Finally, the example above assumed the User Agent supported re-
   INVITE.  If it didn't (i.e. returned a 488 Not Acceptable Here), the
   Application Server would have issued a CANCEL to the VoiceXML 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



Burke, et al.           Expires January 10, 2008               [Page 24]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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]).

   2.  AMR-NB audio [RFC4867] SHOULD be supported.

   3.  MPEG-4 video [RFC3016] SHOULD be supported.

   4.  MPEG-4 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.

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 why 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.





















Burke, et al.           Expires January 10, 2008               [Page 25]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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 a HTTP POST 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
   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).

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
   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



Burke, et al.           Expires January 10, 2008               [Page 26]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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 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.

   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



Burke, et al.           Expires January 10, 2008               [Page 27]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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:

      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































Burke, et al.           Expires January 10, 2008               [Page 28]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


5.  Outbound Calling

   Outbound calls can be triggered via the Application Server using
   either third party call control [RFC3725] or with the SIP REFER
   mechanism [RFC3515].

5.1.  Third Party Call Control Mechanism

   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.  Avoids 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.

5.2.  REFER Mechanism

   The Application Server can place a REFER request to the VoiceXML
   Media Server outside of a SIP dialog to initiate an outbound call.
   The Request-URI in the REFER is constructed identical to that of an
   INVITE to the VoiceXML Media Server and carries the same semantics.
   The Refer-To header contains the URI for the VoiceXML Media Server to
   place the call to.

   On receipt of the REFER request, the VoiceXML Media Server MUST issue
   a provisional response, 100 Trying.  The 202 Accepted response
   indicates that the VoiceXML document has been fetched and parsed
   correctly.  The VoiceXML Media Server proceeds to place the outbound
   INVITE and will execute the application after the ACK is sent.

   If the VoiceXML Session cannot be started, then the VoiceXML Media
   Server MUST respond to the REFER request using the procedure defined
   in section 2.2 above.

   An example is of the REFER initiated outbound call is given below.
   The NOTIFY messages, which contain message/sipfrag bodies [RFC3515],
   allow the Application Server to monitor the progress of the outbound
   call attempt.



Burke, et al.           Expires January 10, 2008               [Page 29]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   Note: An in-dialog REFER will result in a 403 Forbidden response.


   HTTP               VoiceXML               SIP
   Application         Media             Application         User
   Server              Server              Server            Agent
    |                    |                    |                |
    |                    |(1) REFER           |                |
    |                    |<-------------------|                |
    |                    |(2) 100 Trying      |                |
    |                    |------------------->|                |
    |                    |(3) NOTIFY          |                |
    |                    |------------------->|                |
    |                    |(4) 200 OK          |                |
    |                    |<-------------------|                |
    |(5) GET             |                    |                |
    |<-------------------|                    |                |
    |(6) 200 OK [VXML]   |                    |                |
    |------------------->|                    |                |
    |                    |(7) 202 Accepted    |                |
    |                    |------------------->|                |
    |                    |(8) INVITE [offer]                   |
    |                    |------------------------------------>|
    |                    |(9) 200 OK [answer]                  |
    |                    |<------------------------------------|
    |                    |(10) NOTIFY         |                |
    |                    |------------------->|                |
    |                    |(11) 200 OK         |                |
    |                    |<-------------------|                |
    |                    |(12) ACK                             |
    |                    |------------------------------------>|
    |                    |(13) RTP/SRTP                        |
    |                    |.....................................|
    |                    |                                     |

















Burke, et al.           Expires January 10, 2008               [Page 30]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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
   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 bi-directional 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.









Burke, et al.           Expires January 10, 2008               [Page 31]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


    +-------------------------+-----------------------------------+
    | 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                           |
    +-------------------------+-----------------------------------+


   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



Burke, et al.           Expires January 10, 2008               [Page 32]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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.

   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



Burke, et al.           Expires January 10, 2008               [Page 33]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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.


    +-------------------------+-----------------------------------+
    | 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                           |
    +-------------------------+-----------------------------------+


   The 405 Method Not Allowed response can be used by the AS to
   gracefully decline bridge transfers

   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 [SIPEX]) 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



Burke, et al.           Expires January 10, 2008               [Page 34]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   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.

   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 recevied 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



Burke, et al.           Expires January 10, 2008               [Page 35]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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










































Burke, et al.           Expires January 10, 2008               [Page 36]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


7.  Contributors

   The editors gratefully acknowledge the following individuals and
   their companies who contributed to this specification:

      R. J. Auburn (Voxeo)

      Hans Bjurstrom (Hewlett-Packard)

      Dave Burke (Google)

      Emily Candell (Comverse)

      Peter Danielsen (Lucent)

      Brian Frasca (Tellme)

      Jeff Haynie (Hakano)

      Scott McGlashan (Hewlett-Packard)

      Matt Oshry (Tellme)

      Mark Scott (Genesys Telecommunications Laboratories, Inc)

      Rao Surapaneni (Tellme)

























Burke, et al.           Expires January 10, 2008               [Page 37]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


8.  Security Considerations

   Exposing network services with well-known addresses may not be
   desirable.  The VoiceXML Media Server SHOULD authenticate and
   authorize requesting endpoints per local policy.  This is
   particularly important for REFER-initated outbound calls.

   Some applications may choose to transfer confidential information to
   or from the VoiceXML Media Server.  The VoiceXML Media Server SHOULD
   implement the sips: and https: schemes to provide data integrity.

   The VoiceXML Media Server SHOULD use authentication and TLS when
   establishing MRCP control sessions with a MRCPv2 Media Resource
   Server.

   To mitigate against the possibility for denial of service attacks,
   the VoiceXML Media Server SHOULD have local policies such as time-
   limiting VoiceXML application execution.

   The VoiceXML Media Server SHOULD support Secure RTP (SRTP) [RFC3711]
   to provide confidentiality, authentication, and replay protection for
   RTP media streams (including RTCP control traffic).





























Burke, et al.           Expires January 10, 2008               [Page 38]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


9.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.













































Burke, et al.           Expires January 10, 2008               [Page 39]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


10.  Changes since last version:

   o  JSON used for serialization/deserialization of ECMAScript objects

   o  Added description of "delegation model"

   o  Clarified transfer not suitable for use in AS/MS architectures

   o  Added inactive as a permissible value for
      session.connection.protocol.sip.media[x].direction

   o  Clarified that some header / Request-URI parameters can only be
      accessed using the array access mechanism

   o  Minor typographic corrections

   o  Updated references


































Burke, et al.           Expires January 10, 2008               [Page 40]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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.

   [RFC2190]  Zhu, C., "RTP Payload Format for H.263 Video Streams",
              RFC 2190, September 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.

   [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



Burke, et al.           Expires January 10, 2008               [Page 41]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


              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.

   [RFC3960]  Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
              Tone Generation in the Session Initiation Protocol (SIP)",
              RFC 3960, December 2004.

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66,
              RFC 3986, January 2005.

   [RFC4240]  Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network
              Media Services with SIP", RFC 4240, December 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.



Burke, et al.           Expires January 10, 2008               [Page 42]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


              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.

   [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 (work in progress), 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", draft-ietf-speechsc-mrcpv2-12 (work
              in progress), Mar 2007.

   [SIPEX]    Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and
              K. Summers, "Session Initiation Protocol Examples",
              draft-ietf-sipping-service-examples (work in progress),
              July 2005.

   [SIPVXML]  Rosenberg, J., Mataga, P., and D. Ladd, "A SIP Interface
              to VoiceXML Dialog Servers", draft-rosenberg-sip-vxml-00
              (work in progress), July 2001.



Burke, et al.           Expires January 10, 2008               [Page 43]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


   [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, et al.           Expires January 10, 2008               [Page 44]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


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


   Jeff Haynie
   Hakano Inc
   1840 North Creek Circle
   Alpharetta, GA  30004
   USA

   Email: jhaynie@hakano.com


   R.J. Auburn
   Voxeo
   100 East Pine Street #600
   Orlando, FL  32801
   USA

   Email: rj@voxeo.com


   Scott McGlashan
   Hewlett-Packard
   Gustav III:s boulevard 36
   SE-16985 Stockholm
   Sweden

   Email: Scott.McGlashan@hp.com






Burke, et al.           Expires January 10, 2008               [Page 45]


Internet-Draft  SIP Interface to VoiceXML Media Services       July 2007


Full Copyright Statement

   Copyright (C) The IETF Trust (2007).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

   This document and the information contained herein are provided on an
   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
   OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND
   THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS
   OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
   THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.


Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   Intellectual Property Rights or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; nor does it represent that it has
   made any independent effort to identify any such rights.  Information
   on the procedures with respect to rights in RFC documents can be
   found in BCP 78 and BCP 79.

   Copies of IPR disclosures made to the IETF Secretariat and any
   assurances of licenses to be made available, or the result of an
   attempt made to obtain a general license or permission for the use of
   such proprietary rights by implementers or users of this
   specification can be obtained from the IETF on-line IPR repository at
   http://www.ietf.org/ipr.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights that may cover technology that may be required to implement
   this standard.  Please address the information to the IETF at
   ietf-ipr@ietf.org.


Acknowledgment

   Funding for the RFC Editor function is provided by the IETF
   Administrative Support Activity (IASA).





Burke, et al.           Expires January 10, 2008               [Page 46]


Html markup produced by rfcmarkup 1.129b, available from https://tools.ietf.org/tools/rfcmarkup/