SIP Interface to VoiceXML Media Services
RFC 5552
| Document | Type | RFC - Proposed Standard (May 2009) Errata | |
|---|---|---|---|
| Authors | Mark Scott , Dave Burke | ||
| Last updated | 2020-01-21 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | (None) | |
| Document shepherd | (None) | ||
| IESG | IESG state | RFC 5552 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | Robert Sparks | ||
| Send notices to | (None) |
RFC 5552
Network Working Group D. Burke
Request for Comments: 5552 Google
Category: Standards Track M. Scott
Genesys
May 2009
SIP Interface to VoiceXML Media Services
Status of This Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (c) 2009 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents in effect on the date of
publication of this document (http://trustee.ietf.org/license-info).
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document.
Abstract
This document describes a SIP interface to VoiceXML media services.
Commonly, Application Servers controlling Media Servers use this
protocol for pure VoiceXML processing capabilities. This protocol is
an adjunct to the full MEDIACTRL protocol and packages mechanism.
Burke & Scott Standards Track [Page 1]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
Table of Contents
1. Introduction ....................................................3
1.1. Use Cases ..................................................3
1.1.1. IVR Services with Application Servers ...............3
1.1.2. PSTN IVR Service Node ...............................4
1.1.3. 3GPP IMS Media Resource Function (MRF) ..............5
1.1.4. CCXML <-> VoiceXML Interaction ......................6
1.1.5. Other Use Cases .....................................6
1.2. Terminology ................................................7
2. VoiceXML Session Establishment and Termination ..................7
2.1. Service Identification .....................................7
2.2. Initiating a VoiceXML Session .............................10
2.3. Preparing a VoiceXML Session ..............................11
2.4. Session Variable Mappings .................................12
2.5. Terminating a VoiceXML Session ............................15
2.6. Examples ..................................................16
2.6.1. Basic Session Establishment ........................16
2.6.2. VoiceXML Session Preparation .......................17
2.6.3. MRCP Establishment .................................18
3. Media Support ..................................................19
3.1. Offer/Answer ..............................................19
3.2. Early Media ...............................................19
3.3. Modifying the Media Session ...............................21
3.4. Audio and Video Codecs ....................................21
3.5. DTMF ......................................................22
4. Returning Data to the Application Server .......................22
4.1. HTTP Mechanism ............................................22
4.2. SIP Mechanism .............................................23
5. Outbound Calling ...............................................25
6. Call Transfer ..................................................25
6.1. Blind .....................................................26
6.2. Bridge ....................................................27
6.3. Consultation ..............................................29
7. Contributors ...................................................31
8. Acknowledgements ...............................................31
9. Security Considerations ........................................31
10. IANA Considerations ...........................................32
11. References ....................................................32
11.1. Normative References .....................................32
11.2. Informative References ...................................35
Appendix A. Notes on Normative References ........................36
Burke & Scott Standards Track [Page 2]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
1. Introduction
VoiceXML [VXML20], [VXML21] is a World Wide Web Consortium (W3C)
standard for creating audio and video dialogs that feature
synthesized speech, digitized audio, recognition of spoken and dual
tone multi-frequency (DTMF) key input, recording of audio and video,
telephony, and mixed-initiative conversations. VoiceXML allows Web-
based development and content delivery paradigms to be used with
interactive video and voice response applications.
This document describes a SIP [RFC3261] interface to VoiceXML media
services. Commonly, Application Servers controlling media servers
use this protocol for pure VoiceXML processing capabilities. SIP is
responsible for initiating a media session to the VoiceXML media
server and simultaneously triggering the execution of a specified
VoiceXML application. This protocol is an adjunct to the full
MEDIACTRL protocol and packages mechanism.
The interface described here leverages a mechanism for identifying
dialog media services first described in [RFC4240]. The interface
has been updated and extended to support the W3C Recommendation for
VoiceXML 2.0 [VXML20] and VoiceXML 2.1 [VXML21]. A set of commonly
implemented functions and extensions have been specified including
VoiceXML dialog preparation, outbound calling, video media support,
and transfers. VoiceXML session variable mappings have been defined
for SIP with an extensible mechanism for passing application-specific
values into the VoiceXML application. Mechanisms for returning data
to the Application Server have also been added.
1.1. Use Cases
The VoiceXML media service user in this document is generically
referred to as an Application Server. In practice, it is intended
that the interface defined by this document be applicable across a
wide range of use cases. Several intended use cases are described
below.
1.1.1. IVR Services with Application Servers
SIP Application Servers provide services to users of the network.
Typically, there may be several Application Servers in the same
network, each specialized in providing a particular service.
Throughout this specification and without loss of generality, we
posit the presence of an Application Server specialized in providing
Interactive Voice Response (IVR) services. A typical configuration
for this use case is illustrated below.
Burke & Scott Standards Track [Page 3]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
+--------------+
| |
| Application |\
| Server | \
| | \ HTTP
SIP +--------------+ \
/ \ \
+-------------+ / SIP \ +--------------+
| |/ \| |
| SIP | | VoiceXML |
| User Agent | RTP/SRTP | Media Server |
| |=====================| |
+-------------+ +--------------+
Assuming the Application Server also supports HTTP, the VoiceXML
application may be hosted on it and served up via HTTP [RFC2616].
Note, however, that the Web model allows the VoiceXML application to
be hosted on a separate (HTTP) Application Server from the (SIP)
Application Server that interacts with the VoiceXML Media Server via
this specification. It is also possible for a static VoiceXML
application to be stored locally on the VoiceXML Media Server,
leveraging the VoiceXML 2.1 [VXML21] <data> mechanism to interact
with a Web/Application Server when dynamic behavior is required. The
viability of static VoiceXML applications is further enhanced by the
mechanisms defined in Section 2.4, through which the Application
Server can make session-specific information available within the
VoiceXML session context.
The approach described in this document is sometimes termed the
"delegation model" -- the Application Server is essentially
delegating programmatic control of the human-machine interactions to
one or more VoiceXML documents running on the VoiceXML Media Server.
During the human-machine interactions, the Application Server remains
in the signaling path and can respond to results returned from the
VoiceXML Media Server or other external network events.
1.1.2. PSTN IVR Service Node
While this document is intended to enable enhanced use of VoiceXML as
a component of larger systems and services, it is intended that
devices that are completely unaware of this specification remain
capable of invoking VoiceXML services offered by a VoiceXML Media
Server compliant with this document. A typical configuration for
this use case is as follows:
Burke & Scott Standards Track [Page 4]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
+-------------+ SIP +--------------+
| |---------------------| |
| IP/PSTN | | VoiceXML |
| Gateway | RTP/SRTP | Media Server |
| |=====================| |
+-------------+ +--------------+
Note also that beyond the invocation and termination of a VoiceXML
dialog, the semantics defined for call transfers using REFER are
intended to be compatible with standard, existing IP/PSTN (Public
Switched Telephone Network) gateways.
1.1.3. 3GPP IMS Media Resource Function (MRF)
The 3rd Generation Partnership Project (3GPP) IP Multimedia Subsystem
(IMS) [TS23002] defines a Media Resource Function (MRF) used to offer
media processing services such as conferencing, transcoding, and
prompt/collect. The capabilities offered by VoiceXML are ideal for
offering richer media processing services in the context of the MRF.
In this architecture, the interface defined here corresponds to the
"Mr" interface to the MRFC (MRF Controller); the implementation of
this interface might use separated MRFC and MRFP (MRF Processor)
elements (as per the IMS architecture), or might be an integrated MRF
(as is common practice).
+----------+
| App |
| Server |
+----------+
|
| SIP (ISC)
|
+----------+ SIP (Mr) +--------------+
| S-CSCF |---------------| VoiceXML |
| | | MRF |
+----------+ +--------------+
||
|| RTP/SRTP (Mb)
||
The above diagram is highly simplified and shows a subset of nodes
typically involved in MRF interactions. It should be noted that
while the MRF will primarily be used by the Application Server via
the Serving Call Session Control Function (S-CSCF), it is also
possible for calls to be routed directly to the MRF without the
involvement of an Application Server.
Burke & Scott Standards Track [Page 5]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
Although the above is described in terms of the 3GPP IMS
architecture, it is intended that it is also applicable to 3GPP2,
Next Generation Network (NGN), and PacketCable architectures that are
converging with 3GPP IMS standards.
1.1.4. CCXML <-> VoiceXML Interaction
Call Control eXtensible Markup Language (CCXML) 1.0 [CCXML10]
applications provide services mainly through controlling the
interaction between Connections, Conferences, and Dialogs. Although
CCXML is capable of supporting arbitrary dialog environments,
VoiceXML is commonly used as a dialog environment in conjunction with
CCXML applications; CCXML is specifically designed to effectively
support the use of VoiceXML. CCXML 1.0 defines language elements
that allow for Dialogs to be prepared, started, and terminated; it
further allows for data to be returned by the dialog environment, for
call transfers to be requested (by the dialog) and responded to by
the CCXML application, and for arbitrary eventing between the CCXML
application and running dialog application.
The interface described in this document can be used by CCXML 1.0
implementations to control VoiceXML Media Servers. Note, however,
that some CCXML language features require eventing facilities between
CCXML and VoiceXML sessions that go beyond what is defined in this
specification. For example, VoiceXML-controlled call transfers and
mid-dialog, application-defined events cannot be fully realized using
this specification alone. A SIP event package [RFC3265] MAY be used
in addition to this specification to provide extended eventing.
1.1.5. Other Use Cases
In addition to the use cases described in some detail above, there
are a number of other intended use cases that are not described in
detail, such as:
1. Use of a VoiceXML Media Server as an adjunct to an IP-based
Private Branch Exchange / Automatic Call Distributor (PBX/ACD),
possibly to provide voicemail/messaging, automated attendant, or
other capabilities.
2. Invocation and control of a VoiceXML session that provides the
voice modality component in a multimodal system.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
1.2. Terminology
Application Server: A SIP Application Server hosts and executes
services, in particular by terminating SIP sessions on a media
server. The Application Server MAY also act as an HTTP server
[RFC2616] in interactions with media servers.
VoiceXML Media Server: A VoiceXML interpreter including a SIP-based
interpreter context and the requisite media processing
capabilities to support VoiceXML functionality.
VoiceXML Session: A VoiceXML Session is a multimedia session
comprising of at least a SIP User Agent, a VoiceXML Media Server,
the data streams between them, and an executing VoiceXML
application.
VoiceXML Dialog: Equivalent to VoiceXML Session.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
2. VoiceXML Session Establishment and Termination
This section describes how to establish a VoiceXML Session, with or
without preparation, and how to terminate a session. This section
also addresses how session information is made available to VoiceXML
applications.
2.1. Service Identification
The SIP Request-URI is used to identify the VoiceXML media service.
The user part of the SIP Request-URI is fixed to "dialog". This is
done to ensure compatibility with [RFC4240], since this document
extends the dialog interface defined in that specification and
because this convention from [RFC4240] is widely adopted by existing
media servers.
Standardizing the SIP Request-URI including the user part also
improves interoperability between Application Servers and media
servers, and reduces the provisioning overhead that would be required
if use of a media server by an Application Server required an
individually provisioned URI. In this respect, this document (and
[RFC4240]) do not add semantics to the user part, but rather
standardize the way that targets on media servers are provisioned.
Further, since Application Servers -- and not human beings -- are
generally the clients of media servers, issues such as interpretation
and internationalization do not apply.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
Exposing a VoiceXML media service with a well-known address may
enhance the possibility of exploitation: the VoiceXML Media Server is
RECOMMENDED to use standard SIP mechanisms to authenticate endpoints
as discussed in Section 9.
The initial VoiceXML document is specified with the "voicexml"
parameter. In addition, parameters are defined that control how the
VoiceXML Media Server fetches the specified VoiceXML document. The
list of parameters defined by this specification is as follows (note
the parameter names are case-insensitive):
voicexml: URI of the initial VoiceXML document to fetch. This will
typically contain an HTTP URI, but may use other URI schemes, for
example, to refer to local, static VoiceXML documents. If the
"voicexml" parameter is omitted, the VoiceXML Media Server may
select the initial VoiceXML document by other means, such as by
applying a default, or may reject the request.
maxage: Used to set the max-age value of the Cache-Control header in
conjunction with VoiceXML documents fetched using HTTP, as per
[RFC2616]. If omitted, the VoiceXML Media Server will use a
default value.
maxstale: Used to set the max-stale value of the Cache-Control
header in conjunction with VoiceXML documents fetched using HTTP,
as per [RFC2616]. If omitted, the VoiceXML Media Server will use
a default value.
method: Used to set the HTTP method applied in the fetch of the
initial VoiceXML document. Allowed values are "get" or "post"
(case-insensitive). Default is "get".
postbody: Used to set the application/x-www-form-urlencoded encoded
[HTML4] HTTP body for "post" requests (or is otherwise ignored).
ccxml: Used to specify a "JSON value" [RFC4627] that is mapped to
the session.connection.ccxml VoiceXML session variable -- see
Section 2.4.
aai: Used to specify a "JSON value" [RFC4627] that is mapped to the
session.connection.aai VoiceXML session variable -- see
Section 2.4.
Other application-specific parameters may be added to the Request-URI
and are exposed in VoiceXML session variables (see Section 2.4).
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
Formally, the Request-URI for the VoiceXML media service has a fixed
user part "dialog". Seven URI parameters are defined (see the
definition of uri-parameter in Section 25.1 of [RFC3261]).
dialog-param = "voicexml=" vxml-url ; vxml-url follows the URI
; syntax defined in [RFC3986]
maxage-param = "maxage=" 1*DIGIT
maxstale-param = "maxstale=" 1*DIGIT
method-param = "method=" ("get" / "post")
postbody-param = "postbody=" token
ccxml-param = "ccxml=" json-value
aai-param = "aai=" json-value
json-value = false /
null /
true /
object /
array /
number /
string ; defined in [RFC4627]
Parameters of the Request-URI in subsequent re-INVITEs are ignored.
One consequence of this is that the VoiceXML Media Server cannot be
instructed by the Application Server to change the executing VoiceXML
Application after a VoiceXML Session has been started.
Special characters contained in the dialog-param, postbody-param,
ccxml-param, and aai-param values must be URL-encoded ("escaped") as
required by the SIP URI syntax, for example, '?' (%3f), '=' (%3d),
and ';' (%3b). The VoiceXML Media Server MUST therefore unescape
these parameter values before making use of them or exposing them to
running VoiceXML applications. It is important that the VoiceXML
Media Server only unescape the parameter values once since the
desired VoiceXML URI value could itself be URL encoded, for example.
Since some applications may choose to transfer confidential
information, the VoiceXML Media Server MUST support the sips: scheme
as discussed in Section 9.
Informative note: With respect to the postbody-param value, since the
application/x-www-form-urlencoded content itself escapes non-
alphanumeric characters by inserting %HH replacements, the escaping
rules above will result in the '%' characters being further escaped
in addition to the '&' and '=' name/value separators.
Burke & Scott Standards Track [Page 9]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
As an example, the following SIP Request-URI identifies the use of
VoiceXML media services, with
'http://appserver.example.com/promptcollect.vxml' as the initial
VoiceXML document, to be fetched with max-age/max-stale values of
3600s/0s, respectively:
sip:dialog@mediaserver.example.com; \
voicexml=http://appserver.example.com/promptcollect.vxml; \
maxage=3600;maxstale=0
2.2. Initiating a VoiceXML Session
A VoiceXML Session is initiated via the Application Server using a
SIP INVITE. Typically, the Application Server will be specialized in
providing VoiceXML services. At a minimum, the Application Server
may behave as a simple proxy by rewriting the Request-URI received
from the User Agent to a Request-URI suitable for consumption by the
VoiceXML Media Server (as specified in Section 2.1). For example, a
User Agent might present a dialed number:
tel:+1-201-555-0123
that the Application Server maps to a directory assistance
application on the VoiceXML Media Server with a Request-URI of:
sip:dialog@ms1.example.com; \
voicexml=http://as1.example.com/da.vxml
Certain header values in the INVITE message to the VoiceXML Media
Server are mapped into VoiceXML session variables and are specified
in Section 2.4.
On receipt of the INVITE, the VoiceXML Media Server issues a
provisional response, 100 Trying, and commences the fetch of the
initial VoiceXML document. The 200 OK response indicates that the
VoiceXML document has been fetched and parsed correctly and is ready
for execution. Application execution commences on receipt of the ACK
(except if the dialog is being prepared as specified in Section 2.3).
Note that the 100 Trying response will usually be sent on receipt of
the INVITE in accordance with [RFC3261], since the VoiceXML Media
Server cannot in general guarantee that the initial fetch will
complete in less than 200 ms. However, certain implementations may
be able to guarantee response times to the initial INVITE, and thus
may not need to send a 100 Trying response.
As an optimization, prior to sending the 200 OK response, the
VoiceXML Media Server MAY execute the application up to the point of
the first VoiceXML waiting state or prompt flush.
Burke & Scott Standards Track [Page 10]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
A VoiceXML Media Server, like any SIP User Agent, may be unable to
accept the INVITE request for a variety of reasons. For instance, a
Session Description Protocol (SDP) offer contained in the INVITE
might require the use of codecs that are not supported by the Media
Server. In such cases, the Media Server should respond as defined by
[RFC3261]. However, there are error conditions specific to VoiceXML,
as follows:
1. If the Request-URI does not conform to this specification, a 400
Bad Request MUST be returned (unless it is used to select other
services not defined by this specification).
2. If a URI parameter in the Request-URI is repeated, then the
request MUST be rejected with a 400 Bad Request response.
3. If the Request-URI does not include a "voicexml" parameter, and
the VoiceXML Media Server does not elect to use a default page,
the VoiceXML Media Server MUST return a final response of 400 Bad
Request, and it SHOULD include a Warning header with a 3-digit
code of 399 and a human-readable error message.
4. If the VoiceXML document cannot be fetched or parsed, the
VoiceXML Media Server MUST return a final response of 500 Server
Internal Error and SHOULD include a Warning header with a 3-digit
code of 399 and a human-readable error message.
Informative note: Certain applications may pass a significant amount
of data to the VoiceXML dialog in the form of Request-URI parameters.
This may cause the total size of the INVITE request to exceed the MTU
of the underlying network. In such cases, applications/
implementations must take care either to use a transport appropriate
to these larger messages (such as TCP) or to use alternative means of
passing the required information to the VoiceXML dialog (such as
supplying a unique session identifier in the initial VoiceXML URI and
later using that identifier as a key to retrieve data from the HTTP
server).
2.3. Preparing a VoiceXML Session
In certain scenarios, it is beneficial to prepare a VoiceXML Session
for execution prior to running it. A previously prepared VoiceXML
Session is expected to execute with minimal delay when instructed to
do so.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
If a media-less SIP dialog is established with the initial INVITE to
the VoiceXML Media Server, the VoiceXML application will not execute
after receipt of the ACK. To run the VoiceXML application, the
Application Server (AS) must issue a re-INVITE to establish a media
session.
A media-less SIP dialog can be established by sending an SDP
containing no media lines in the initial INVITE. Alternatively, if
no SDP is sent in the initial INVITE, the VoiceXML Media Server will
include an offer in the 200 OK message, which can be responded to
with an answer in the ACK with the media port(s) set to 0.
Once a VoiceXML application is running, a re-INVITE that disables the
media streams (i.e., sets the ports to 0) will not otherwise affect
the executing application (except that recognition actions initiated
while the media streams are disabled will result in noinput
timeouts).
2.4. Session Variable Mappings
The standard VoiceXML session variables are assigned values according
to:
session.connection.local.uri: Evaluates to the SIP URI specified in
the To: header of the initial INVITE.
session.connection.remote.uri: Evaluates to the SIP URI specified in
the From: header of the initial INVITE.
session.connection.redirect: This array is populated by information
contained in the History-Info [RFC4244] header in the initial
INVITE or is otherwise undefined. Each entry (hi-entry) in the
History-Info header is mapped, in reverse order, into an element
of the session.connection.redirect array. Properties of each
element of the array are determined as follows:
* uri - Set to the hi-targeted-to-uri value of the History-Info
entry
* pi - Set to 'true' if hi-targeted-to-uri contains a
"Privacy=history" parameter, or if the INVITE Privacy header
includes 'history'; 'false' otherwise
* si - Set to the value of the "si" parameter if it exists,
undefined otherwise
* reason - Set verbatim to the value of the "Reason" parameter of
hi-targeted-to-uri
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
session.connection.protocol.name: Evaluates to "sip". Note that
this is intended to reflect the use of SIP in general, and does
not distinguish between whether the media server was accessed via
SIP or SIPS procedures.
session.connection.protocol.version: Evaluates to "2.0".
session.connection.protocol.sip.headers: This is an associative
array where each key in the array is the non-compact name of a SIP
header in the initial INVITE converted to lowercase (note the case
conversion does not apply to the header value). If multiple
header fields of the same field name are present, the values are
combined into a single comma-separated value. Implementations
MUST at a minimum include the Call-ID header and MAY include other
headers. For example,
session.connection.protocol.sip.headers["call-id"] evaluates to
the Call-ID of the SIP dialog.
session.connection.protocol.sip.requesturi: This is an associative
array where the array keys and values are formed from the URI
parameters on the SIP Request-URI of the initial INVITE. The
array key is the URI parameter name converted to lowercase (note
the case conversion does not apply to the parameter value). The
corresponding array value is obtained by evaluating the URI
parameter value as a "JSON value" [RFC4627] in the case of the
ccxml-param and aai-param values and otherwise as a string. In
addition, the array's toString() function returns the full SIP
Request-URI. For example, assuming a Request-URI of sip:dialog@
example.com;voicexml=http://example.com;aai=%7b"x":1%2c"y":true%7d
then session.connection.protocol.sip.requesturi["voicexml"]
evaluates to "http://example.com",
session.connection.protocol.sip.requesturi["aai"].x evaluates to 1
(type Number), session.connection.protocol.sip.requesturi["aai"].y
evaluates to true (type Boolean), and
session.connection.protocol.sip.requesturi evaluates to the
complete Request-URI (type String) 'sip:dialog@
example.com;voicexml=http://example.com;aai={"x":1,"y":true}'.
session.connection.aai: Evaluates to
session.connection.protocol.sip.requesturi["aai"].
session.connection.ccxml: Evaluates to
session.connection.protocol.sip.requesturi["ccxml"].
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
session.connection.protocol.sip.media: This is an array where each
array element is an object with the following properties:
* type: - This required property indicates the type of the media
associated with the stream. The value is a string. It is
strongly recommended that the following values are used for
common types of media: "audio" for audio media, and "video" for
video media.
* direction: - This required property indicates the
directionality of the media relative to
session.connection.originator. Defined values are sendrecv,
sendonly, recvonly, and inactive.
* format: - This property is optional. If defined, the value of
the property is an array. Each array element is an object that
specifies information about one format of the media (there is
an array element for each payload type on the m-line). The
object contains at least one property called "name" whose value
is the MIME subtype of the media format (MIME subtypes are
registered in [RFC4855]). Other properties may be defined with
string values; these correspond to required and, if defined,
optional parameters of the format.
As a consequence of this definition, there is an array entry in
session.connection.protocol.sip.media for each non-disabled m-line
for the negotiated media session. Note that this session variable
is updated if the media session characteristics for the VoiceXML
Session change (i.e., due to a re-INVITE). For an example,
consider a connection with bidirectional G.711 mu-law "audio"
sampled at 8 kHz. In this case,
session.connection.protocol.sip.media[0].type evaluates to
"audio", session.connection.protocol.sip.media[0].direction to
"sendrecv",
session.connection.protocol.sip.media[0].format[0].name evaluates
to "audio/PCMU", and
session.connection.protocol.sip.media[0].format[0].rate evaluates
to "8000".
Note that when accessing SIP headers and Request-URI parameters via
the session.connection.protocol.sip.headers and
session.connection.protocol.sip.requesturi associative arrays defined
above, applications can choose between two semantically equivalent
ways of referring to the array. For example, either of the following
can be used to access a Request-URI parameter named "foo":
session.connection.protocol.sip.requesturi["foo"]
session.connection.protocol.sip.requesturi.foo
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
However, it is important to note that not all SIP header names or
Request-URI parameter names are valid ECMAScript identifiers, and as
such, can only be accessed using the first form (array notation).
For example, the Call-ID header can only be accessed as
session.connection.protocol.sip.headers["call-id"]; attempting to
access the same value as
session.connection.protocol.sip.headers.call-id would result in an
error.
2.5. Terminating a VoiceXML Session
The Application Server can terminate a VoiceXML Session by issuing a
BYE to the VoiceXML Media Server. Upon receipt of a BYE in the
context of an existing VoiceXML Session, the VoiceXML Media Server
MUST send a 200 OK response and MUST throw a
'connection.disconnect.hangup' event to the VoiceXML application. If
the Reason header [RFC3326] is present on the BYE Request, then the
value of the Reason header is provided verbatim via the '_message'
variable within the catch element's anonymous variable scope.
The VoiceXML Media Server may also initiate termination of the
session by issuing a BYE request. This will typically occur as a
result of encountering a <disconnect> or <exit> in the VoiceXML
application, due to the VoiceXML application running to completion,
or due to unhandled errors within the VoiceXML application.
See Section 4 for mechanisms to return data to the Application
Server.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
2.6. Examples
2.6.1. Basic Session Establishment
This example illustrates an Application Server setting up a VoiceXML
Session on behalf of a User Agent.
SIP VoiceXML HTTP
User Application Media Application
Agent Server Server Server
| | | |
|(1) INVITE [offer] | | |
|------------------->|(2) INVITE [offer] | |
|(3) 100 Trying |------------------->| |
|<-------------------|(4) 100 Trying | |
| |<-------------------| |
| | | |
| | |(5) GET |
| | |------------------->|
| | |(6) 200 OK [VXML] |
| | |<-------------------|
| | | |
| |(7) 200 OK [answer] | |
|(8) 200 OK [answer] |<-------------------| |
|<-------------------| | |
|(9) ACK | | |
|------------------->|(10) ACK | |
| |------------------->| (execute |
|(11) RTP/SRTP | | VoiceXML |
|.........................................| application) |
| | | |
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
2.6.2. VoiceXML Session Preparation
This example demonstrates the preparation of a VoiceXML Session. In
this example, the VoiceXML session is prepared prior to placing an
outbound call to a User Agent, and is started as soon as the User
Agent answers.
The [answer1:0] notation is used to indicate an SDP answer with the
media ports set to 0.
SIP VoiceXML HTTP
User Application Media Application
Agent Server Server Server
| | | |
| |(1) INVITE | |
| |-------------------->| |
| |(2) 100 Trying | |
| |<--------------------| |
| | | |
| | |(3) GET |
| | |------------------->|
| | |(4) 200 OK [VXML] |
| | |<-------------------|
| | | |
| |(5) 200 OK [offer1] | |
| |<--------------------| |
| |(6) ACK [answer1:0] | |
|(7) INVITE |-------------------->| |
|<-------------------| | |
|(8) 200 OK [offer2] | | |
|------------------->|(9) INVITE [offer2'] | |
| |-------------------->| |
| |(10) 100 Trying | |
| |<--------------------| |
| |(11) 200 OK [answer2]| |
|(12) ACK [answer2] |<--------------------| |
|<-------------------|(13) ACK | |
| |-------------------->| (execute |
|(14) RTP/SRTP | VoiceXML |
|..........................................| application) |
| | | |
Implementation detail: offer2' is derived from offer2 -- it
duplicates the m-lines and a-lines from offer2. However, offer2'
differs from offer2 since it must contain the same o-line as used in
answer1:0 but with the version number incremented. Also, if offer1
has more m-lines than offer2, then offer2' must be padded with extra
(rejected) m-lines.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
2.6.3. MRCP Establishment
Media Resource Control Protocol (MRCP) [MRCPv2] is a protocol that
enables clients such as a VoiceXML Media Server to control media
service resources such as speech synthesizers, recognizers,
verifiers, and identifiers residing in servers on the network.
The example below illustrates how a VoiceXML Media Server may
establish an MRCP session in response to an initial INVITE.
VoiceXML HTTP
User Media MRCPv2 Application
Agent Server Server Server
| | | |
|(1) INVITE [offer1] | | |
|------------------->| | |
|(2) 100 Trying | | |
|<-------------------|(3) GET | |
| |---------------------------------------->|
| | | |
| |(4) 200 OK [VXML] | |
| |<----------------------------------------|
| | | |
| |(5) INVITE [offer2] | |
| |--------------------->| |
| | | |
| |(6) 200 OK [answer2] | |
| |<---------------------| |
| | | |
| |(7) ACK | |
| |--------------------->| |
| | | |
| |(8) MRCP connection | |
| |<-------------------->| |
|(9) 200 OK [answer1]| | |
|<-------------------| | |
| | | |
|(10) ACK | | |
|------------------->| | |
| | | |
|(11) RTP/SRTP | | |
|...........................................| |
| | | |
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
In this example, the VoiceXML Media Server is responsible for
establishing a session with the MRCPv2 Media Resource Server prior to
sending the 200 OK response to the initial INVITE. The VoiceXML
Media Server will perform the appropriate offer/answer with the
MRCPv2 Media Resource Server based on the SDP capabilities of the
Application Server and the MRCPv2 Media Resource Server. The
VoiceXML Media Server will change the offer received from step 1 to
establish an MRCPv2 session in step (5) and will re-write the SDP to
include an m-line for each MRCPv2 resource to be used and other
required SDP modifications as specified by MRCPv2. Once the VoiceXML
Media Server performs the offer/answer with the MRCPv2 Media Resource
Server, it will establish an MRCPv2 control channel in step (8). The
MRCPv2 resource is deallocated when the VoiceXML Media Server
receives or sends a BYE (not shown).
3. Media Support
This section describes the mandatory and optional media support
required by this interface.
3.1. Offer/Answer
The VoiceXML Media Server MUST support the standard offer/answer
mechanism of [RFC3264]. In particular, if an SDP offer is not
present in the INVITE, the VoiceXML Media Server will make an offer
in the 200 OK response listing its supported codecs.
3.2. Early Media
The VoiceXML Media Server MAY support early establishment of media
streams as described in [RFC3960]. This allows the Application
Server to establish media streams between a User Agent and the
VoiceXML Media Server in parallel with the initial VoiceXML document
being processed (which may involve dynamic VoiceXML page generation
and interaction with databases or other systems). This is useful
primarily for minimizing the delay in starting a VoiceXML Session,
particularly in cases where a session with the User Agent already
exists but the media stream associated with that session needs to be
redirected to a VoiceXML Media Server.
The following flow demonstrates the use of early media (using the
Gateway model defined in [RFC3960]):
Burke & Scott Standards Track [Page 19]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
SIP VoiceXML HTTP
User Application Media Application
Agent Server Server Server
| | | |
|..(existing session)..| | |
| |(1) INVITE | |
| |------------------>| |
| | |(2) HTTP GET |
| | |------------------>|
| |(3) 183 [offer] | |
|(4) re-INVITE [offer] |<------------------| |
|<---------------------| | |
|(5) 200 OK [answer] | | |
|--------------------->| | |
|(6) ACK | | |
|<---------------------| | |
| | (7) PRACK [answer]| |
| |------------------>| |
| | (8) PRACK 200 OK | |
| |<------------------| |
|(9) RTP/SRTP | | |
|..........................................| |
| | |(10) 200 OK [VXML] |
| | |<------------------|
| | | |
| |(11) 200 OK | |
| |<------------------| |
| |(12) ACK | |
| |------------------>| (execute |
| | | VoiceXML |
| | | application) |
| | | |
Although [RFC3960] prefers the use of the Application Server model
for early media over the Gateway model, the primary issue with the
Gateway model -- forking -- is significantly less common when issuing
requests to VoiceXML Media Servers. This is because VoiceXML Media
Servers respond to all requests with 200 OK responses in the absence
of unusual errors, and they typically do so within several hundred
milliseconds. This makes them unlikely targets in forking scenarios,
since alternative targets of the forking process would virtually
never be able to respond more quickly than an automated system,
unless they are themselves automated systems -- in which case, there
is little point in setting up a response time race between two
automated systems. Issues with ringing tone generation in the
Gateway model are also mitigated, both by the typically quick 200 OK
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
response time, and because this specification mandates that no media
packets are generated until the receipt of an ACK (thus eliminating
the need for the User Agent to perform media packet analysis).
Note that the offer of early media by a VoiceXML Media Server does
not imply that the referenced VoiceXML application can always be
fetched and executed successfully. For instance, if the HTTP
Application Server were to return a 4xx response in step 10 above, or
if the provided VoiceXML content was not valid, the VoiceXML Media
Server would still return a 500 response (as per Section 2.2). At
this point, it would be the responsibility of the Application Server
to tear down any media streams established with the media server.
3.3. Modifying the Media Session
The VoiceXML Media Server MUST allow the media session to be modified
via a re-INVITE and SHOULD support the UPDATE method [RFC3311] for
the same purpose. In particular, it MUST be possible to change
streams between sendrecv, sendonly, and recvonly as specified in
[RFC3264].
Unidirectional streams are useful for announcement- or listening-only
(hotword). The preferred mechanism for putting the media session on
hold is specified in [RFC3264], i.e., the UA modifies the stream to
be sendonly and mutes its own stream. Modification of the media
session does not affect VoiceXML application execution (except that
recognition actions initiated while on hold will result in noinput
timeouts).
3.4. Audio and Video Codecs
For the purposes of achieving a basic level of interoperability, this
section specifies a minimal subset of codecs and RTP [RFC3550]
payload formats that MUST be supported by the VoiceXML Media Server.
For audio-only applications, G.711 mu-law and A-law MUST be supported
using the RTP payload type 0 and 8 [RFC3551]. Other codecs and
payload formats MAY be supported.
Video telephony applications, which employ a video stream in addition
to the audio stream, are possible in VoiceXML 2.0/2.1 through the use
of multimedia file container formats such as the .3gp [TS26244] and
.mp4 formats [IEC14496-14]. Video support is optional for this
specification. If video is supported then:
1. H.263 Baseline [RFC4629] MUST be supported. For legacy reasons,
the 1996 version of H.263 MAY be supported using the RTP payload
format defined in [RFC2190] (payload type 34 [RFC3551]).
Burke & Scott Standards Track [Page 21]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
2. Adaptive Multi-Rate (AMR) narrow band audio [RFC4867] SHOULD be
supported.
3. MPEG-4 video [RFC3016] SHOULD be supported.
4. MPEG-4 Advanced Audio Coding (AAC) audio [RFC3016] SHOULD be
supported.
5. Other codecs and payload formats MAY be supported.
Video record operations carried out by the VoiceXML Media Server
typically require receipt of an intra-frame before the recording can
commence. The VoiceXML Media Server SHOULD use the mechanism
described in [RFC4585] to request that a new intra-frame be sent.
Since some applications may choose to transfer confidential
information, the VoiceXML Media Server MUST support Secure RTP (SRTP)
[RFC3711] as discussed in Section 9.
3.5. DTMF
DTMF events [RFC4733] MUST be supported. When the User Agent does
not indicate support for [RFC4733], the VoiceXML Media Server MAY
perform DTMF detection using other means such as detecting DTMF tones
in the audio stream. Implementation note: the reason only [RFC4733]
telephone-events must be used when the User Agent indicates support
of it is to avoid the risk of double detection of DTMF if detection
on the audio stream was simultaneously applied.
4. Returning Data to the Application Server
This section discusses the mechanisms for returning data (e.g.,
collected utterance or digit information) from the VoiceXML Media
Server to the Application Server.
4.1. HTTP Mechanism
At any time during the execution of the VoiceXML application, data
can be returned to the Application Server via HTTP using standard
VoiceXML elements such as <submit> or <subdialog>. Notably, the
<data> element in VoiceXML 2.1 [VXML21] allows data to be sent to the
Application Server efficiently without requiring a VoiceXML page
transition and is ideal for short VoiceXML applications such as
"prompt and collect".
For most applications, it is necessary to correlate the information
being passed over HTTP with a particular VoiceXML Session. One way
this can be achieved is to include the SIP Call-ID (accessible in
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
VoiceXML via the session.connection.protocol.sip.headers array)
within the HTTP POST fields. Alternatively, a unique "POST-back URI"
can be specified as an application-specific URI parameter in the
Request-URI of the initial INVITE (accessible in VoiceXML via the
session.connection.protocol.sip.requesturi array).
Since some applications may choose to transfer confidential
information, the VoiceXML Media Server MUST support the https: scheme
as discussed in Section 9.
4.2. SIP Mechanism
Data can be returned to the Application Server via the expr or
namelist attribute on <exit> or the namelist attribute on
<disconnect>. A VoiceXML Media Server MUST support encoding of the
expr/namelist data in the message body of a BYE request sent from the
VoiceXML Media Server as a result of encountering the <exit> or
<disconnect> element. A VoiceXML Media Server MAY support inclusion
of the expr/namelist data in the message body of the 200 OK message
in response to a received BYE request (i.e., when the VoiceXML
application responds to the connection.disconnect.hangup event and
subsequently executes an <exit> element with the expr or namelist
attribute specified).
Note that sending expr/namelist data in the 200 OK response requires
that the VoiceXML Media Server delay the final response to the
received BYE request until the VoiceXML application's post-disconnect
final processing state terminates. This mechanism is subject to the
constraint that the VoiceXML Media Server must respond before the
User Agent Client's (UAC's) timer F expires (defaults to 32 seconds).
Moreover, for unreliable transports, the UAC will retransmit the BYE
request according to the rules of [RFC3261]. The VoiceXML Media
Server SHOULD implement the recommendations of [RFC4320] regarding
when to send the 100 Trying provisional response to the BYE request.
If a VoiceXML application executes a <disconnect> [VXML21] and then
subsequently executes an <exit> with namelist information, the
namelist information from the <exit> element is discarded.
Namelist variables are first converted to their "JSON value"
equivalent [RFC4627] and encoded in the message body using the
application/x-www-form-urlencoded format content type [HTML4]. The
behavior resulting from specifying a recording variable in the
namelist or an ECMAScript object with circular references is not
defined. If the expr attribute is specified on the <exit> element
instead of the namelist attribute, the reserved name __exit is used.
Burke & Scott Standards Track [Page 23]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
To allow the Application Server to differentiate between a BYE
resulting from a <disconnect> from one resulting from an <exit>, the
reserved name __reason is used, with a value of "disconnect" (without
brackets) to reflect the use of VoiceXML's <disconnect> element, and
a value of "exit" (without brackets) to an explicit <exit> in the
VoiceXML document. If the session terminates for other reasons (such
as the media server encountering an error), this parameter may be
omitted, or may take on platform-specific values prefixed with an
underscore.
This specification extends the application/x-www-form-urlencoded by
replacing non-ASCII characters with one or more octets of the UTF-8
representation of the character, with each octet in turn replaced by
%HH, where HH represents the uppercase hexadecimal notation for the
octet value and % is a literal character. As a consequence, the
Content-Type header field in a BYE message containing expr/namelist
data MUST be set to application/x-www-form-urlencoded;charset=utf-8.
The following table provides some examples of <exit> usage and the
corresponding result content.
+----------------------------------------------------------------+
|<exit> Usage | Result Content |
|------------------------------|---------------------------------|
|<exit/> | __reason=exit |
|<exit expr="5"/> | __exit=5&__reason=exit |
|<exit expr="'done'"/> | __exit="done"&__reason=exit |
|<exit expr="userAuthorized"/> | __exit=true&__reason=exit |
|<exit namelist="pin errors"/> | pin=1234&errors=0&__reason=exit |
+----------------------------------------------------------------+
assuming the following VoiceXML variables and values:
userAuthorized = true
pin = 1234
errors = 0
For example, consider the VoiceXML snippet:
...
<exit namelist="id pin"/>
...
If id equals 1234 and pin equals 9999, say, the BYE message would
look similar to:
Burke & Scott Standards Track [Page 24]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
BYE sip:user@pc33.example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10
Max-Forwards: 70
From: sip:dialog@example.com;tag=a6c85cf
To: sip:user@example.com;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 231 BYE
Content-Type: application/x-www-form-urlencoded;charset=utf-8
Content-Length: 30
id=1234&pin=9999&__reason=exit
Since some applications may choose to transfer confidential
information, the VoiceXML Media Server MUST support the S/MIME
encoding of SIP message bodies as discussed in Section 9.
5. Outbound Calling
Outbound calls can be triggered via the Application Server using
third-party call control [RFC3725].
Flow IV from [RFC3725] is recommended in conjunction with the
VoiceXML Session preparation mechanism. This flow has several
advantages over others, namely:
1. Selection of a VoiceXML Media Server and preparation of the
VoiceXML application can occur before the call is placed to avoid
the callee experiencing delays.
2. Avoidance of timing difficulties that could occur with other
flows due to the time taken to fetch and parse the initial
VoiceXML document.
3. The flow is IPv6 compatible.
An example flow for an Application-Server-initiated outbound call is
provided in Section 2.6.2.
6. Call Transfer
While VoiceXML is at its core a dialog language, it also provides
optional call transfer capability. VoiceXML's transfer capability is
particularly suited to the PSTN IVR Service Node use case described
in Section 1.1.2. It is NOT RECOMMENDED to use VoiceXML's call
transfer capability in networks involving Application Servers.
Rather, the Application Server itself can provide call routing
Burke & Scott Standards Track [Page 25]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
functionality by taking signaling actions based on the data returned
to it from the VoiceXML Media Server via HTTP or in the SIP BYE
message.
If VoiceXML transfer is supported, the mechanism described in this
section MUST be employed. The transfer flows specified here are
selected on the basis that they provide the best interworking across
a wide range of SIP devices. CCXML<->VoiceXML implementations, which
require tight-coupling in the form of bidirectional eventing to
support all transfer types defined in VoiceXML, may benefit from
other approaches, such as the use of SIP event packages [RFC3265].
In what follows, the provisional responses have been omitted for
clarity.
6.1. Blind
The blind-transfer sequence is initiated by the VoiceXML Media Server
via a REFER message [RFC3515] on the original SIP dialog. The
Refer-To header contains the URI for the called party, as specified
via the dest or destexpr attributes on the VoiceXML <transfer> tag.
If the REFER request is accepted, in which case the VoiceXML Media
Server will receive a 2xx response, the VoiceXML Media Server throws
the connection.disconnect.transfer event and will terminate the
VoiceXML Session with a BYE message. For blind transfers,
implementations MAY use [RFC4488] to suppress the implicit
subscription associated with the REFER message.
If the REFER request results in a non-2xx response, the <transfer>'s
form item variable (or event raised) depends on the SIP response and
is specified in the following table. Note that this indicates that
the transfer request was rejected.
+-------------------------+-----------------------------------+
| SIP Response | <transfer> variable / event |
+-------------------------+-----------------------------------+
| 404 Not Found | error.connection.baddestination |
| 405 Method Not Allowed | error.unsupported.transfer.blind |
| 503 Service Unavailable | error.connection.noresource |
| (No response) | network_busy |
| (Other 3xx/4xx/5xx/6xx) | unknown |
+-------------------------+-----------------------------------+
Burke & Scott Standards Track [Page 26]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
An example is illustrated below (provisional responses and NOTIFY
messages corresponding to provisional responses have been omitted for
clarity).
User Agent 1 VoiceXML User Agent 2
(Caller) Media Server (Callee)
| | |
|(0) RTP/SRTP | |
|.................| |
| | |
|(1) REFER | <transfer> |
|<----------------| |
|(2) 202 Accepted | |
|---------------->| |
|(3) BYE | |
|<----------------| |
|(4) 200 OK | |
|---------------->| |
| | Stop RTP (0) |
|(5) INVITE |
|---------------------------------->|
|(6) 200 OK |
|<----------------------------------|
|(7) NOTIFY | |
|---------------->| |
|(8) 200 OK | |
|<--------------- | |
|(9) ACK |
|---------------------------------->|
|(10) RTP/SRTP |
|...................................|
| | |
If the aai or aaiexpr attribute is present on <transfer>, it is
appended to the Refer-To URI as a parameter named "aai" in the REFER
method. Reserved characters are URL-encoded as required for SIP/SIPS
URIs [RFC3261]. The mapping of values outside of the ASCII range is
platform specific.
6.2. Bridge
The bridge transfer function results in the creation of a small
multi-party session involving the Caller, the VoiceXML Media Server,
and the Callee. The VoiceXML Media Server invites the Callee to the
session and will eject the Callee if the transfer is terminated.
Burke & Scott Standards Track [Page 27]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
If the aai or aaiexpr attribute is present on <transfer>, it is
appended to the Request-URI in the INVITE as a URI parameter named
"aai". Reserved characters are URL-encoded as required for SIP/SIPS
URIs [RFC3261]. The mapping of values outside of the ASCII range is
platform specific.
During the transfer attempt, audio specified in the transferaudio
attribute of <transfer> is streamed to User Agent 1. A VoiceXML
Media Server MAY play early media received from the Callee to the
Caller if the transferaudio attribute is omitted.
The bridge transfer sequence is illustrated below. The VoiceXML
Media Server (acting as a UAC) makes a call to User Agent 2 with the
same codecs used by User Agent 1. When the call setup is complete,
RTP flows between User Agent 2 and the VoiceXML Media Server. This
stream is mixed with User Agent 1's.
User Agent 1 VoiceXML User Agent 2
(Caller) Media Server (Callee)
| | |
|(0)RTP/SRTP | |
|...................| |
| | |
| <transfer>|(1)INVITE [offer] |
| |------------------>|
| |(2) 200 OK [answer]|
| |<------------------|
| |(3) ACK |
| |------------------>|
| |(4) RTP/SRTP |
| mix |...................|
| (0)+(4)| |
If a final response is not received from User Agent 2 from the INVITE
and the connecttimeout expires (specified as an attribute of
<transfer>), the VoiceXML Media Server will issue a CANCEL to
terminate the transaction and the <transfer>'s form item variable is
set to noanswer.
If INVITE results in a non-2xx response, the <transfer>'s form item
variable (or event raised) depends on the SIP response and is
specified in the following table.
Burke & Scott Standards Track [Page 28]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
+-------------------------+-----------------------------------+
| SIP Response | <transfer> variable / event |
+-------------------------+-----------------------------------+
| 404 Not Found | error.connection.baddestination |
| 405 Method Not Allowed | error.unsupported.transfer.bridge |
| 408 Request Timeout | noanswer |
| 486 Busy Here | busy |
| 503 Service Unavailable | error.connection.noresource |
| (No response) | network_busy |
| (Other 3xx/4xx/5xx/6xx) | unknown |
+-------------------------+-----------------------------------+
Once the transfer is established, the VoiceXML Media Server can
"listen" to the media stream from User Agent 1 to perform speech or
DTMF hotword, which when matched results in a near-end disconnect,
i.e., the VoiceXML Media Server issues a BYE to User Agent 2 and the
VoiceXML application continues with User Agent 1. A BYE will also be
issued to User Agent 2 if the call duration exceeds the maximum
duration specified in the maxtime attribute on <transfer>.
If User Agent 2 issues a BYE during the transfer, the transfer
terminates and the VoiceXML <transfer>'s form item variable receives
the value far_end_disconnect. If User Agent 1 issues a BYE during
the transfer, the transfer terminates and the VoiceXML event
connection.disconnect.transfer is thrown.
6.3. Consultation
The consultation transfer (also called attended transfer [RFC5359])
is similar to a blind transfer except that the outcome of the
transfer call setup is known and the Caller is not dropped as a
result of an unsuccessful transfer attempt.
Consultation transfer commences with the same flow as for bridge
transfer except that the RTP streams are not mixed at step (4) and
error.unsupported.transfer.consultation supplants
error.unsupported.transfer.bridge. Assuming a new SIP dialog with
User Agent 2 is created, the remainder of the sequence follows as
illustrated below (provisional responses and NOTIFY messages
corresponding to provisional responses have been omitted for
clarity). Consultation transfer makes use of the Replaces: header
[RFC3891] such that User Agent 1 calls User Agent 2 and replaces the
latter's SIP dialog with the VoiceXML Media Server with a new SIP
dialog between the Caller and Callee.
Burke & Scott Standards Track [Page 29]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
User Agent 1 VoiceXML User Agent 2
(Caller) Media Server (Callee)
| | |
|(0) RTP/SRTP | |
|.................|(4) RTP/SRTP |
| |.................|
|(5) REFER | |
|<----------------| |
|(6) 202 Accepted | |
|---------------->| |
|(7) INVITE Replaces:ms1.example.com|
|---------------------------------->|
|(8) 200 OK |
|<----------------------------------|
|(9) ACK |
|---------------------------------->|
|(10) RTP/SRTP |
|...................................|
| |(11) BYE |
| |<----------------|
| |(12) 200 OK |
| |---------------->| Stop
|(13) NOTIFY | | RTP (4)
|---------------->| |
|(14) 200 OK | |
|<----------------| |
|(15) BYE | |
|<----------------| |
|(16) 200 OK | |
|---------------->| Stop |
| | RTP (0) |
If a response other than 202 Accepted is received in response to the
REFER request sent to User Agent 1, the transfer terminates and an
error.unsupported.transfer.consultation event is raised. In
addition, a BYE is sent to User Agent 2 to terminate the established
outbound leg.
The VoiceXML Media Server uses receipt of a NOTIFY message with a
sipfrag message of 200 OK to determine that the consultation transfer
has succeeded. When this occurs, the connection.disconnect.transfer
event will be thrown to the VoiceXML application, and a BYE is sent
to User Agent 1 to terminate the session. A NOTIFY message with a
non-2xx final response sipfrag message body will result in the
transfer terminating and the associated VoiceXML input item variable
being set to 'unknown'. Note that as a consequence of this
Burke & Scott Standards Track [Page 30]
RFC 5552 SIP Interface to VoiceXML Media Services May 2009
mechanism, implementations MUST NOT use [RFC4488] to suppress the
implicit subscription associated with the REFER message for
consultation transfers.
7. Contributors
The bulk of the early work for this effort was carried out on weekly
teleconferences and over email. The authors would particularly like
to recognize the contributions of R. J. Auburn (Voxeo), Jeff Haynie
(Hakano), and Scott McGlashan (Hewlett-Packard).
8. Acknowledgements
This document owes its genesis to, "A SIP Interface to VoiceXML
Dialog Servers", authored by J. Rosenberg, P. Mataga, and D. Ladd.
The following people had input to the current document:
R. J. Auburn (Voxeo)
Hans Bjurstrom (Hewlett-Packard)
Emily Candell (Comverse)
Peter Danielsen (Lucent)
Brian Frasca (Tellme)
Jeff Haynie (Hakano)
Scott McGlashan (Hewlett-Packard)
Matt Oshry (Tellme)
Rao Surapaneni (Tellme)
The authors would like to acknowledge the support of Cullen Jennings
and the Mediactrl chairs, Eric Burger and Spencer Dawkins.
9. Security Considerations
Exposing a VoiceXML media service with a well-known address may
enhance the possibility of exploitation (for example, an invoked
network service may trigger a billing event). The VoiceXML Media
Server is RECOMMENDED to use standard SIP mechanisms [RFC3261] to
authenticate requesting endpoints and authorize per local policy.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
Some applications may choose to transfer confidential information to
or from the VoiceXML Media Server. To provide data confidentiality,
the VoiceXML Media Server MUST implement the sips: and https: schemes
in addition to S/MIME message body encoding as described in
[RFC3261].
The VoiceXML Media Server MUST support Secure RTP (SRTP) [RFC3711] to
provide confidentiality, authentication, and replay protection for
RTP media streams (including RTCP control traffic).
To mitigate the possibility of denial-of-service attacks, the
VoiceXML Media Server is RECOMMENDED (in addition to authenticating
and authorizing endpoints described above) to provide mechanisms for
implementing local policies such as the time-limiting of VoiceXML
application execution.
10. IANA Considerations
IANA has registered the following parameters in the SIP/SIPS URI
Parameters registry, following the Specification Required policy of
[RFC3969]:
Parameter Name Predefined Values Reference
-------------- ----------------- ---------
maxage No RFC 5552
maxstale No RFC 5552
method "get" / "post" RFC 5552
postbody No RFC 5552
ccxml No RFC 5552
aai No RFC 5552
11. References
11.1. Normative References
[HTML4] Raggett, D., Le Hors, A., and I. Jacobs, "HTML 4.01
Specification", W3C Recommendation, Dec 1999.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee,
"Hypertext Transfer Protocol -- HTTP/1.1", RFC 2616,
June 1999.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
[RFC3016] Kikuchi, Y., Nomura, T., Fukunaga, S., Matsui, Y., and
H. Kimata, "RTP Payload Format for MPEG-4 Audio/Visual
Streams", RFC 3016, November 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G.,
Johnston, A., Peterson, J., Sparks, R., Handley, M.,
and E. Schooler, "SIP: Session Initiation Protocol",
RFC 3261, June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
Model with Session Description Protocol (SDP)",
RFC 3264, June 2002.
[RFC3265] Roach, A., "Session Initiation Protocol (SIP)-Specific
Event Notification", RFC 3265, June 2002.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP)
UPDATE Method", RFC 3311, October 2002.
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The
Reason Header Field for the Session Initiation
Protocol (SIP)", RFC 3326, December 2002.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP)
Refer Method", RFC 3515, April 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio
and Video Conferences with Minimal Control", STD 65,
RFC 3551, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
K. Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.
[RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
Camarillo, "Best Current Practices for Third Party
Call Control (3pcc) in the Session Initiation Protocol
(SIP)", BCP 85, RFC 3725, April 2004.
[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session
Initiation Protocol (SIP) "Replaces" Header",
RFC 3891, September 2004.
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RFC 5552 SIP Interface to VoiceXML Media Services May 2009
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter,
"Uniform Resource Identifier (URI): Generic Syntax",
STD 66, RFC 3986, January 2005.
[RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information",
RFC 4244, November 2005.
[RFC4320] Sparks, R., "Actions Addressing Identified Issues with
the Session Initiation Protocol's (SIP) Non-INVITE
Transaction", RFC 4320, January 2006.
[RFC4488] Levin, O., "Suppression of Session Initiation Protocol
(SIP) REFER Method Implicit Subscription", RFC 4488,
May 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J.
Rey, "Extended RTP Profile for Real-time Transport
Control Protocol (RTCP)-Based Feedback (RTP/AVPF)",
RFC 4585, July 2006.
[RFC4627] Crockford, D., "The application/json Media Type for
JavaScript Object Notation (JSON)", RFC 4627,
July 2006.
[RFC4629] Ott, H., Bormann, C., Sullivan, G., Wenger, S., and R.
Even, "RTP Payload Format for ITU-T Rec", RFC 4629,
January 2007.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals",
RFC 4733, December 2006.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q.
Xie, "RTP Payload Format and File Storage Format for
the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
Wideband (AMR-WB) Audio Codecs", RFC 4867, April 2007.
[VXML20] McGlashan, S., Burnett, D., Carter, J., Danielsen, P.,
Ferrans, J., Hunt, A., Lucas, B., Porter, B., Rehor,
K., and S. Tryphonas, "Voice Extensible Markup
Language (VoiceXML) Version 2.0", W3C Recommendation,
March 2004.
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[VXML21] Oshry, M., Auburn, R J., Baggia, P., Bodell, M.,
Burke, D., Burnett, D., Candell, E., Kilic, H.,
McGlashan, S., Lee, A., Porter, B., and K. Rehor,
"Voice Extensible Markup Language (VoiceXML) Version
2.1", W3C Candidate Recommendation, June 2005.
11.2. Informative References
[CCXML10] Auburn, R J., "Voice Browser Call Control: CCXML
Version 1.0", W3C Working Draft, June 2005.
[IEC14496-14] "Information technology. Coding of audio-visual
objects. MP4 file format", ISO/IEC ISO/IEC 14496-
14:2003, October 2003.
[MRCPv2] Shanmugham, S. and D. Burnett, "Media Resource Control
Protocol Version 2 (MRCPv2)", Work in Progress,
November 2008.
[RFC2190] Zhu, C., "RTP Payload Format for H.263 Video Streams",
RFC 2190, September 1997.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and
Ringing Tone Generation in the Session Initiation
Protocol (SIP)", RFC 3960, December 2004.
[RFC3969] Camarillo, G., "The Internet Assigned Number Authority
(IANA) Uniform Resource Identifier (URI) Parameter
Registry for the Session Initiation Protocol (SIP)",
BCP 99, RFC 3969, December 2004.
[RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic
Network Media Services with SIP", RFC 4240,
December 2005.
[RFC5359] Johnston, A., Sparks, R., Cunningham, C., Donovan, S.,
and K. Summers, "Session Initiation Protocol Service
Examples", BCP 144, RFC 5359, October 2008.
[TS23002] "3rd Generation Partnership Project: Network
architecture (Release 6)", 3GPP TS 23.002 v6.6.0,
December 2004.
[TS26244] "Transparent end-to-end packet switched streaming
service (PSS); 3GPP file format (3GP)", 3GPP TS 26.244
v6.4.0, December 2004.
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Appendix A. Notes on Normative References
We make a "downref" normative reference to [RFC4627] -- an
Informational document describing a proprietary (but extremely
popular) format.
Authors' Addresses
Dave Burke
Google
Belgrave House, 76 Buckingham Palace Road
London SW1W 9TQ
United Kingdom
EMail: daveburke@google.com
Mark Scott
Genesys
1120 Finch Avenue West, 8th floor
Toronto, Ontario M3J 3H7
Canada
EMail: Mark.Scott@genesyslab.com
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