D. Burke
Internet-Draft Voxpilot
Expires: June 11, 2006 M. Scott
VoiceGenie
J. Haynie
Vocalocity
R. Auburn
Voxeo
S. McGlashan
Hewlett-Packard
December 8, 2005
SIP Interface to VoiceXML Media Services
draft-burke-vxml-00.txt
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Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
This document describes a SIP interface to VoiceXML media services,
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which is commonly employed between application servers and media
servers offering VoiceXML processing capabilities.
Comments
Comments are solicited and should be addressed to the authors.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1.1. IVR Services with Application Servers . . . . . . . . 4
1.1.2. PSTN IVR Service Node . . . . . . . . . . . . . . . . 5
1.1.3. 3GPP IMS Media Resource Function (MRF) . . . . . . . . 6
1.1.4. CCXML <-> VoiceXML Interaction . . . . . . . . . . . . 6
1.1.5. Other Use Cases . . . . . . . . . . . . . . . . . . . 7
1.2. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7
2. VoiceXML Session Establishment and Termination . . . . . . . . 8
2.1. Service Identification . . . . . . . . . . . . . . . . . . 8
2.2. Initiating a VoiceXML Session . . . . . . . . . . . . . . 9
2.3. Preparing a VoiceXML Session . . . . . . . . . . . . . . . 11
2.4. Terminating a VoiceXML Session . . . . . . . . . . . . . . 11
2.5. Session Variable Mappings . . . . . . . . . . . . . . . . 11
2.6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . 13
2.6.1. Basic Session Establishment . . . . . . . . . . . . . 13
2.6.2. VoiceXML Session Preparation . . . . . . . . . . . . . 14
2.6.3. MRCP Establishment . . . . . . . . . . . . . . . . . . 15
3. Media Support . . . . . . . . . . . . . . . . . . . . . . . . 18
3.1. Offer/Answer . . . . . . . . . . . . . . . . . . . . . . . 18
3.2. Early Media . . . . . . . . . . . . . . . . . . . . . . . 18
3.3. Modifying the Media Session . . . . . . . . . . . . . . . 19
3.4. Audio and Video Codecs . . . . . . . . . . . . . . . . . . 19
3.5. DTMF . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
4. Returning Data to the Application Server . . . . . . . . . . . 21
4.1. HTTP Mechanism . . . . . . . . . . . . . . . . . . . . . . 21
4.2. SIP Mechanism . . . . . . . . . . . . . . . . . . . . . . 21
5. Outbound Calling . . . . . . . . . . . . . . . . . . . . . . . 23
5.1. Third Party Call Control Mechanism . . . . . . . . . . . . 23
5.2. REFER Mechanism . . . . . . . . . . . . . . . . . . . . . 23
6. Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . 25
6.1. Blind . . . . . . . . . . . . . . . . . . . . . . . . . . 25
6.2. Bridge . . . . . . . . . . . . . . . . . . . . . . . . . . 27
6.3. Consultation . . . . . . . . . . . . . . . . . . . . . . . 29
7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 31
8. Security Considerations . . . . . . . . . . . . . . . . . . . 32
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 34
10.1. Normative References . . . . . . . . . . . . . . . . . . . 34
10.2. Informative References . . . . . . . . . . . . . . . . . . 36
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 37
Intellectual Property and Copyright Statements . . . . . . . . . . 38
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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 [NETANN]. 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
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.
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+--------------+
| |
| Application |\
| Server | \
| | \ HTTP
SIP +--------------+ \
/ \ \
+-------------+ / SIP \ +--------------+
| |/ \| |
| SIP | | VoiceXML |
| User Agent | RTP | Media Server |
| |=====================| |
+-------------+ +--------------+
Consistent with the Web model, the VoiceXML application may reside
directly on the Application Server and is served up via HTTP
[RFC2616]. Note, however, that the application is not required to
reside on a single Application Server since 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 2.5, through
which the Application Server can make session-specific information
available within the VoiceXML session context.
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 [NETANN] 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:
+-------------+ SIP +--------------+
| |---------------------| |
| IP/PSTN | | VoiceXML |
| Gateway | RTP | Media Server |
| |=====================| |
+-------------+ +--------------+
Note also that beyond the invocation and termination of a VoiceXML
dialog, the semantics defined for call transfers using REFER are
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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 (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, filter criteria on the S-CSCF could route calls directly
to the MRF independently of Application Servers.
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.
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
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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.
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.
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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 [NETANN]. 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: URL 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 2.5).
Parameters of the Request-URI in subsequent re-INVITEs are ignored.
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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.
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
unencode Request-URI parameter values before making use of them or
exposing them to running VoiceXML applications.
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:8965
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
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 2.5.
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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 MUST 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.
As an optimization, prior to 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
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 the use of an HTTP redirector). This note also
applies if the dialog is started using a REFER request as described
in section 5.2.
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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.
2.4. 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 the VoiceXML Media
Server MUST send a 200 OK response, and MUST throw a
'connection.disconnect.hangup' event to the VoiceXML application.
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.
2.5. 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).
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session.connection.redirect: This array is populated by information
contained in the Diversion [DIV] header in the initial INVITE or
is otherwise undefined. Each URI entry in the Diversion header is
mapped, in reverse order, into an element of the
session.connection.redirect array. Properties of each element of
the array are mapped according to: the name-addr value is mapped
to the uri property, the privacy parameter is mapped to the pi
property, and the screen parameter is mapped to the si property.
The reason parameter in the Diversion header is mapped to the
reason property according to:
* unknown - "unknown",
* user-busy - "user busy",
* no-answer - "no reply",
* deflection - "deflection immediate response",
* unavailable - "mobile subscriber not reachable".
Other values of the reason parameter in the Diversion header are
mapped verbatim to the reason property.
session.connection.protocol.name: Evaluates to "sip".
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 case-sensitive, non-compact
name of a SIP header in the initial INVITE. 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 the 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)
according to the following rules:
* If the URI parameter name includes no periods, the key (of type
string) is formed from the entire parameter name and its
corresponding value is of type string and evaluates to the URI
parameter value.
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* If the URI parameter name includes a period, the array key (of
type string) is formed from the characters to the left of the
period and the corresponding value is of type object. A
property is added to that object whose name is formed from the
characters to the right of the period (up to the next period).
If the URI parameter name contains no further periods, the
property is of type string and evaluates to the URI parameter
value. Otherwise it is of type object and the process of
adding properties repeats.
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;obj.y=2;obj.z.a=3
then session.connection.protocol.requesturi["voicexml"] evaluates
to "http://ajax.com",
session.connection.protocol.requesturi["obj"].x evaluates to "1",
session.connection.protocol.requesturi["obj"].y evaluates to "2",
session.connection.protocol.requesturi["obj"].z.a evaluates to
"3", session.connection.protocol.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.codecs: This is an array where each
array element corresponds to a codec currently in use by the
VoiceXML Session. Each element in the array is an object with at
least one property called name. The name property evaluates to
the MIME type [RFC3555] of the codec in use. Required parameters
for a codec (and any optional parameters present) are mapped to
corresponding named string properties. For example, for a media
session employing G.711 mu-law audio sampled at 8kHz,
session.connection.protocol.sip.codecs[0].name evaluates to
"audio/PCMU" and session.connection.protocol.sip.codecs[0].rate
evaluates to "8000". Note that this session variable is updated
if the codecs for the VoiceXML Session change (due to a re-
INVITE).
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.
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User Agent Application Server VoiceXML Media Server
| | |
|(1) INVITE [offer] | |
|----------------------->|(2) INVITE [offer] |
|(3) 100 Trying |----------------------->|
|<-----------------------|(4) 100 Trying |
| |<-----------------------|
| | |
| |(5) HTTP GET | (fetch
| |<-----------------------| initial
| |(6) HTTP 200 OK [VXML] | VoiceXML)
| |----------------------->| document)
| | |
| |(7) 200 OK [answer] |
|(8) 200 OK [answer] |<-----------------------|
|<-----------------------| |
|(9) ACK | |
|----------------------->|(10) ACK |
| |----------------------->| (execute
|(11) RTP | | 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.
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User Agent Application Server VoiceXML Media Server
| | |
| |(1) INVITE |
| |----------------------->|
| |(2) 100 Trying |
| |<-----------------------|
| | |
| |(3) HTTP GET | (fetch
| |<-----------------------| initial
| |(4) HTTP 200 OK [VXML] | VoiceXML)
| |----------------------->| document)
| | |
| |(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 | VoiceXML
|.................................................| application)
| | |
2.6.3. MRCP Establishment
The VoiceXML Media Server SHOULD use the [MRCPv2] protocol to handle
media processing resources for speech recognition, speech synthesis,
speaker verification and speaker identification.
The example illustrates a VoiceXML Media Server establishing an MRCP
Session.
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Application Server VoiceXML MS MRCPv2 Server Web Server
| | | |
|(1) INVITE [offer1] | | |
|------------------->| | |
|(2) 100 Trying | | |
|<-------------------|(3) HTTP GET | |
| |---------------------------------------->|
| | | |
| |(4) HTTP 200 OK [VXML]| |
| |<----------------------------------------|
| | | |
| |(5) INVITE [offer2] | |
| |--------------------->| |
| | | |
| |(6) 200 OK [answer2] | |
| |<---------------------| |
| | | |
| |(7) ACK | |
| |--------------------->| |
| | | |
| |(8) MRCP connection | |
| |<-------------------->| |
|(9) 200 OK [answer1]| | |
|<-------------------| | |
| | | |
|(10) ACK | | |
|------------------->| | |
| | | |
|(11) RTP | | |
...............................................| |
| | | |
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).
If a media-less SIP dialog is established with the initial INVITE to
the VoiceXML Media Server, a MRCP session MUST not be established
until the Application Server issues a re-invite to the VoiceXML Media
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Server.
If an MRCP resource is required and an MRCP session cannot be
established, the VoiceXML Media Server MUST decline service by
returning a 503 Temporarily Unavailable final response. The VoiceXML
Media Server SHOULD include a Warning header with a 3-digit code of
399 and a human readable error message such as "recognizer resource
unavailable".
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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:
User Agent Application Server VoiceXML Media Server
| | |
|...(Existing session)...| |
| |(1) INVITE |
| |----------------------->|
| |(2) 183 [offer] |
|(3) re-INVITE [offer] |<-----------------------|
|<-----------------------| |
|(4) 200 OK [answer] | |
|----------------------->| |
|(5) ACK | |
|<-----------------------| |
| |(6) HTTP GET | (fetch
| |<-----------------------| initial
| |(7) HTTP 200 OK [VXML] | VoiceXML)
| |----------------------->| document)
| | |
| |(8) 200 OK [offer] |
| |<-----------------------|
| |(9) ACK [answer] |
| |----------------------->| (execute
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|(10) RTP | | VoiceXML
|.................................................| application)
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. As such, Application
Servers may choose to ignore any early media SDP from 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.
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 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
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.mp4 formats [IEC14496-14]. Video support is optional for this
specification. If video is supported then:
1. H.263 Baseline [RFC2429] 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 [RFC3267] 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.
DTMF events [RFC2833] MUST be supported. The VoiceXML Media Server
MAY perform DTMF detection using other means such as detecting DTMF
within the audio stream. If the SDP from the user agent indicates
support for [RFC2833] telephone-event then that mechanism SHOULD be
used only.
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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 namelist
attribute on <exit> or <disconnect>. Namelist variables are first
converted to a string (via the ECMAScript toString() operation) and
encoded in the message body of the BYE message using the application/
x-www-form-urlencoded format content type [HTML4]. The behavior of
including a recording variable in the namelist is not defined.
Note: This mechanism relies on a BYE being issued from the VoiceXML
Media Server and hence is only available when the VoiceXML
Application terminates. While this feature is useful for many
applications (e.g. prompt and collect), the HTTP mechanism, which
allows for mid-call information to be sent to the Application Server,
may be preferable for some applications.
If a VoiceXML Application returns data using the namelist attribute
on <disconnect> [VXML21] and subsequently employs a namelist
attribute on <exit>, the latter namelist information is discarded.
This specification extends the application/x-www-form-urlencoded by
replacing non-ASCII characters with one or more octets of the the
UTF-8 representation of the character, with each octet in turn
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replaced by %HH, where HH represents the uppercase hexadecimal
notation for the octet value and % is a literal character. The
content type MUST include the charset parameter to indicate UTF-8
encoding.
For example, consider the VoiceXML snippet:
...
<exit namelist="id pin"/>
...
If id equals 1234 and pin equals 0000, 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: 16
id=1234&pin=0000
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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.
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Note: An in-dialog REFER will result in a 403 Forbidden response.
VoiceXML Media Server Application Server User Agent
| | |
|(1) REFER | |
|<-----------------------| |
|(2) 100 Trying | |
|----------------------->| |
|(3) NOTIFY | |
|----------------------->| |
|(4) 200 OK | |
|<-----------------------| |
|(5) HTTP GET | |
|----------------------->| |
|(6) HTTP 200 OK [VXML] | |
|<-----------------------| |
|(7) 202 Accepted | |
|----------------------->| |
|(8) INVITE [offer] |
|------------------------------------------------>|
|(9) 200 OK [answer] |
|<------------------------------------------------|
|(10) NOTIFY | |
|----------------------->| |
|(11) 200 OK | |
|<-----------------------| |
|(12) ACK |
|------------------------------------------------>|
|(13) RTP |
|.................................................|
| |
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6. Call Transfer
Transfer capability is optional in VoiceXML [VXML20], [VXML21]. If
transfer is supported, it MUST use the mechanism described in this
section.
The flows specified here assume that the Application Server is
assuming a proxy server role. More complex behaviors are possible,
for example the Application Server could act as a B2BUA to ensure it
remains in the signaling path during transfers where the VoiceXML
Media Server has dropped out. Note that the mechanisms used by the
VoiceXML Media Server are still valid in the absence of an
Application Server. In what follows, the provisional responses have
been omitted for clarity.
Note: 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].
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 part, as specified
via the 'dest' or 'destexpr' attributes on the VoiceXML <transfer>
tag.
If the REFER request succeeds, in which case the VoiceXML Media
Server will receive a 202 Accepted, the VoiceXML Media Server throws
the connection.disconnect.transfer event and will terminate the
VoiceXML Session with a BYE 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, and does not indicate the outcome
of actually performing the transfer (e.g. busy, no answer, etc).
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+-------------------------+-----------------------------------+
| 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 (NOTIFY messages and provisional
responses have been omitted for clarity). In this example, the
Application Server behaves as a proxy and is not in the signaling
path of the transferred call.
User Agent 1 Application VoiceXML User Agent 2
(Caller) Server Media Server (Callee)
| | | |
|(0) RTP | | |
|...................................| |
| | | |
| |(1) REFER | |
|(2) REFER |<----------------| <transfer> |
|<----------------| | |
|(3) 202 Accepted | | |
|---------------->|(4) 202 Accepted | |
| |---------------->| |
| |(5) BYE | |
|(6) BYE |<----------------| |
|<----------------| | |
|(7) 200 OK | | |
|---------------->|(8) 200 OK | |
| |---------------->| Stop RTP (0) |
|(9) INVITE |
|---------------------------------------------------->|
|(10) 200 OK |
|<----------------------------------------------------|
|(11) ACK |
|---------------------------------------------------->|
|(13) RTP |
|.....................................................|
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.
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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.
Issue Note: There is currently no standard mechanism to allow the
Application Server to indicate dynamically to the VoiceXML Media
Server that it wishes to be included in the signaling path for the
new outbound SIP dialog. One possible mechanism is to add a
specific parameter to the Application Server's URI in the Record-
Route header in the initial inbound INVITE. When present this
could be used to indicate that the VoiceXML Media Server must add
that URI to its pre-exiting route set for outbound INVITEs. In
the flows illustrated below, it is assumed that the Application
Server is in the pre-existing route set configured in the local
policy.
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". 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 via the Application Server. 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.
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User Agent 1 Application VoiceXML User Agent 2
(Caller) Server Media Server (Callee)
| | | |
|(0)RTP | | |
|.......................................| |
| | | |
| |(1)INVITE [offer] | <transfer> |
| |<------------------| |
| |(2)INVITE [offer] | |
| |-------------------------------------->|
| |(4) 200 OK [answer] |
| |<--------------------------------------|
| |(5) 200 OK [answer]| |
| |------------------>| |
| |(6) ACK | |
| |<------------------| |
| |(7) ACK |
| |-------------------------------------->|
| | |(8) RTP |
| | |...................| mix
| | | | 0+8
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.
+-------------------------+-----------------------------------+
| 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
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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 (8) 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. Consultation transfer makes use of the Replaces:
header such that User Agent 1 calls User Agent 2 and replaces the
latter's SIP dialog with the VoiceXML Media Server with a new SIP
dialog between the Caller and Callee.
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User Agent 1 Application VoiceXML User Agent 2
(Caller) Server Media Server (Callee)
| | | |
|(0) RTP | | |
|...................................|(8) RTP |
| | |.................|
| |(9) REFER | |
|(10) REFER |<----------------| |
|<----------------| | |
|(11) 202 Accepted| | |
|---------------->|(12) 202 Accepted| |
| |---------------->| |
|(13) INVITE Replaces: ms1.example.com |
|---------------------------------------------------->|
|(14) 200 OK |
|<----------------------------------------------------|
|(15) ACK |
|---------------------------------------------------->|
|(16) RTP |
|.....................................................|
| | |(17) BYE |
| | |<----------------|
| | |(18) 200 OK |
| | |---------------->| Stop
|(19) NOTIFY | | | RTP (8)
|---------------->|(20) NOTIFY | |
| |---------------->| |
| |(21) 200 OK | |
|(22) 200 OK |<----------------| |
|<----------------| | |
| |(23) BYE | |
|(24) BYE |<----------------| |
|<----------------| | |
|(25) 200 OK | | |
|---------------->|(26) 200 OK | | Stop
| |---------------->| | RTP (0)
| | | |
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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 (Voxpilot)
Emily Candell (Comverse)
Brian Frasca (Tellme)
Jeff Haynie (Vocalocity)
Scott McGlashan (Hewlett-Packard)
Mark Scott (VoiceGenie)
Rao Surapaneni (Tellme)
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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.
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9. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
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10. References
10.1. Normative References
[DIV] Levy, S. and B. Byerly, "Diversion Indication in SIP",
draft-levy-sip-diversion-08 (work in progress),
August 2004.
[HTML4] Raggett, D., Le Hors, A., and I. Jacobs, "HTML 4.01
Specification", W3C Recommendation, Dec 1999.
[MRCPv2] Shanmugham, S. and D. Burnett, "Media Resource Control
Protocol Version 2", draft-ietf-speechsc-mrcpv2-09 (work
in progress), Oct 2005.
[NETANN] Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network
Media Services with SIP", draft-burger-sipping-netann-11
(work in progress), February 2005.
[RFC1890] Schulzrinne, H., "RTP Profile for Audio and Video
Conferences with Minimal Control", RFC 1890, January 1996.
[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.
[RFC2429] Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco,
C., Newell, D., Ott, J., Sullivan, G., Wenger, S., and C.
Zhu, "RTP Payload Format for the 1998 Version of ITU-T
Rec. H.263 Video (H.263+)", RFC 2429, October 1998.
[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.
[RFC2833] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF
Digits, Telephony Tones and Telephony Signals", RFC 2833,
May 2000.
[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,
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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.
[RFC3267] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"Real-Time Transport Protocol (RTP) Payload Format and
File Storage Format for the Adaptive Multi-Rate (AMR) and
Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs",
RFC 3267, June 2002.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP)
UPDATE Method", RFC 3311, October 2002.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3555] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Formats", RFC 3555, July 2003.
[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.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[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.
Burke, et al. Expires June 11, 2006 [Page 35]
Internet-Draft SIP Interface to VoiceXML Media Services December 2005
10.2. 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.
[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.
[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 June 11, 2006 [Page 36]
Internet-Draft SIP Interface to VoiceXML Media Services December 2005
Authors' Addresses
Dave Burke
Voxpilot
6 - 9 Trinity Street
Dublin 2
Ireland
Email: david.burke@voxpilot.com
Mark Scott
VoiceGenie
1120 Finch Avenue West, 8th floor
Toronto, Ontario M3J 3H7
Canada
Email: mscott@voicegenie.com
Jeff Haynie
Vocalocity
730 Peachtree Street, Suite 1100
Atlanta, GA 30308
USA
Email: jhaynie@vocalocity.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 June 11, 2006 [Page 37]
Internet-Draft SIP Interface to VoiceXML Media Services December 2005
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Burke, et al. Expires June 11, 2006 [Page 38]
