Speechsc D. Oran
Internet-Draft Cisco Systems, Inc.
Expires: December 5, 2003 June 6, 2003
Requirements for Distributed Control of ASR, SI/SV and TTS Resources
draft-ietf-speechsc-reqts-05
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Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
This document outlines the needs and requirements for a protocol to
control distributed speech processing of audio streams. By speech
processing, this document specifically means automatic speech
recognition (ASR), speaker recognition - which includes both speaker
identification (SI) and speaker verification (SV) - and
text-to-speech (TTS). Other IETF protocols, such as SIP and RTSP,
address rendezvous and control for generalized media streams.
However, speech processing presents additional requirements that none
of the extant IETF protocols address.
1. Introduction
There are multiple IETF protocols for establishment and termination
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of media sessions (SIP [5]), low-level media control (MGCP [6] and
MEGACO [7]), and media record and playback (RTSP [8]). This document
focuses on requirements for one or more protocols to support the
control of network elements that perform Automated Speech Recognition
(ASR), speaker identification or verification (SI/SV), and rendering
text into audio, also known as Text-to-Speech (TTS). Many multimedia
applications can benefit from having automatic speech recognition
(ASR) and text-to-speech (TTS) processing available as a distributed,
network resource. This requirements document limits its focus to the
distributed control of ASR, SI/SV and TTS servers.
There are a broad range of systems which can benefit from a unified
approach to control of TTS, ASR, and SI/SV. These include
environments such as VoIP gateways to the PSTN, IP Telephones, media
servers, and wireless mobile devices who obtain speech services via
servers on the network.
To date, there are a number of proprietary ASR and TTS API's, as well
as two IETF drafts that address this problem [12], [13]. However,
there are serious deficiencies to the existing drafts. In
particular, they mix the semantics of existing protocols yet are
close enough to other protocols as to be confusing to the
implementer.
This document sets forth requirements for protocols to support
distributed speech processing of audio streams. For simplicity, and
to remove confusion with existing protocol proposals, this document
presents the requirements as being for a "framework" that addresses
the distributed control of speech resources It refers to such a
framework as "SPEECHSC", for Speech Services Control.
Discussion of this and related documents is on the speechsc mailing
list. To subscribe, send the message "subscribe speechsc" to
speechsc-request@ietf.org. The public archive is at http://
www.ietf.org/mail-archive/workinggroups/speechsc/current/
maillist.html
2. SPEECHSC Framework
Figure 1 below shows the SPEECHSC framework for speech processing.
+-------------+
| Application |
| Server |\
+-------------+ \ SPEECHSC
SIP, VoiceXML, / \
etc. / \
+------------+ / \ +-------------+
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| Media |/ SPEECHSC \---| ASR, SI/SV |
| Processing |-------------------------| and/or TTS |
RTP | Entity | RTP | Server |
=====| |=========================| |
+------------+ +-------------+
Figure 1: Figure 1: SPEECHSC Framework
The "Media Processing Entity" is a network element that processes
media. It may be either a pure media handler, or also have an
associated SIP user agent, VoiceXML browser or other control entity.
The "ASR, SI/SV and/or TTS Server" is a network element which
performs the back-end speech processing. It may generate an RTP
stream as output based on text input (TTS) or return recognition
results in response to an RTP stream as input (ASR, SI/SV). The
"Application Server" is a network element that instructs the Media
Processing Entity on what transformations to make to the media
stream. Those instructions may be established via a session protocol
such as SIP, or provided via a client/server exchange such as
VoiceXML. The framework allows either the Media Processing Entity or
the Application Server to control the ASR or TTS Server using
SPEECHSC as a control protocol, which accounts for the speechsc
protocol appearing twice in the diagram.
Physical embodiments of the entities can reside in one physical
instance per entity, or some combination of entities. For example, a
VoiceXML [10] Gateway may combine the ASR and TTS functions on the
same platform as the Media Processing Entity. Note that VoiceXML
Gateways themselves are outside the scope of this protocol. Likewise,
one can combine the Application Server and Media Processing Entity,
as would be the case in an interactive voice response (IVR) platform.
One can also decompose the Media Processing Entity into an entity
that controls media endpoints and entities that process media
directly. Such would be the case with a decomposed gateway using
MGCP or megaco. However, this decomposition is again orthogonal to
the scope of SPEECHSC. The following subsections provide a number of
example use cases the SPEECHSC, one each for TTS, ASR and SI/SV. They
are intended to be illustrative only, and not to imply any
restriction on the scope of the framework or to limit the
decompostion or configuration to that shown in the example.
2.1 TTS Example
This example illustrates a simple usage of SPEECHSC to provide a Text
to Speech service for playing announcements to a user on a phone with
no display for textual error messages. The example scenario is shown
below in figure 2. In the figure, the VoIP gateway acts as both the
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Media Processing Entity and the Application Server of the SPEECHSC
framework in figure 1.
+---------+
_| SIP |
_/ | Server |
+-----------+ SIP/ +---------+
| | _/
+-------+ | VoIP |_/
| POTS |___| Gateway | RTP +---------+
| Phone | | (SIP UA) |=========| |
+-------+ | |\_ | SPEECHSC|
+-----------+ \ | TTS |
\__ | Server |
SPEECHSC | |
\_| |
+---------+
Figure 2: Figure 2: Text to speech example of SPEECHSC
The POTS phone on the left attempts to make a phone call. The VoIP
gateway, acting as a SIP UA, tries to establish a SIP session to
complete the call, but gets an error, such as a SIP "486 Busy Here"
response. Without SPEECHSC the gateway would most likely just output
a busy signal to the POTS phone, However, with SPEECHSC access to a
TTS server it can provide a spoken error message. The VoIP Gateway
therefore constructs a text error string using information from the
SIP messages, such as "Your call to 978-555-1212 did not go through
because the called party was busy". It then can use SPEECHSC to
establish an association with a SPEECHSC server, open an RTP stream
between itself and the server, and issue a TTS request for the error
message, which will be played to the user on the POTS phone.
2.2 Automatic speech recognition example
This example illustrates a VXML-enabled media processing entity and
associated application server using the SPEECHSC framework to supply
an ASR-based user interface through an Interactive Voice Response
(IVR) system. The example scenario is shown below in figure 3. The
VXML-client corresponds to the "media processing entity", while the
IVR application server corresponds to the "application server" of
the SPEECHSC framework of figure 1.
+------------+
| IVR |
_|Application |
VXML_/ +------------+
+-----------+ __/
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| |_/ +------------+
PSTN Trunk | VoIP | SPEECHSC| |
=============| Gateway |---------| SPEECHSC |
|(VXML voice| | ASR |
| browser) |=========| Server |
+-----------+ RTP +------------+
Figure 3: Figure 3: Automatic speech recognition example
In this example, users call into the service in order to obtain stock
quotes. The VoIP gateway answers their PSTN call. An IVR
application feeds VXML scripts to the gateway to drive the user
interaction. The VXML interpreter on the gateway directs the user's
media stream to the SPEECHSC ASR server and uses SPEECHSC to control
the ASR server.
When, for example, the user speaks the name of a stock in response to
an IVR prompt, the SPEECHSC ASR server attempts recognition of the
name, and returns the results to the VXML gateway. The VXML gateway,
following standard VXML mechanisms, informs the IVR Application of
the recognized result. The IVR Application can then do the
appropriate information lookup. The answer, of course, can be sent
back to the user using text-to-speech. This example does not show
this scenario, but it would work analogously to the scenario shown in
section Section 2.1.
2.3 Speaker Identification example
This example illustrates using speaker identification to allow
voice-actuated login to an IP phone. The example scenario is shown
below in figure 4. In the figure, the IP Phone acts as both the
"Media Processing Entity" and the Application Server" of the SPEECHSC
framework in figure 1.
+-----------+ +---------+
| | RTP | |
| IP |=========| SPEECHSC|
| Phone | | TTS |
| |_________| Server |
| | SPEECHSC| |
+-----------+ +---------+
Figure 4: Figure 4: Speaker identification example
In this example, a user speaks into a SIP phone in order to get
"logged in" to that phone to make and receive phone calls using his
identity and preferences. The IP phone uses the SPEECHSC framework to
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set up an RTP stream between the phone and the SPEECHSC SI/SV server
and to request verification. The SV server verifies the user's
identity and returns the result, including the necessary login
credentials, to the phone via SPEECHSC. The IP Phone may either use
the identity directly to identify the user in outgoing calls, to
fetch the user's preferences from a configuration server, request
authorization from a AAA server, in any combination. Since this
example uses SPEECHSC to perform a security-related function, be sure
to note the associated material in Section 9
3. General Requirements
3.1 Reuse Existing Protocols
To the extent feasible, the SPEECHSC framework SHOULD use existing
protocols.
3.2 Maintain Existing Protocol Integrity
In meeting the requirement of Section 3.1, the SPEECHSC framework
MUST NOT redefine the semantics of an existing protocol. Said
differently, we will not break existing protocols or cause backward
compatibility problems.
3.3 Avoid Duplicating Existing Protocols
To the extent feasible, SPEECHSC SHOULD NOT duplicate the
functionality of existing protocols. For example, network
announcements using SIP [11] and RTSP [8] already define how to
request playback of audio. The focus of SPEECHSC is new functionality
not addressed by existing protocols or extending existing protocols
within the strictures of the requirement in Section 3.2. Where an
existing protocol can be gracefully extended to support SPEECHSC
requirements, such extensions are acceptable alternatives for meeting
the requirements.
As a corollary to this, the SPEECHSC should not require a separate
protocol to perform functions that could be easily added into the
SPEECHSC protocol (like redirecting media streams, or discovering
capabilities), unless it is similarly easy to embed that protocol
directly into the SPEECHSC framework.
3.4 Efficiency
The SPEECHSC framework SHOULD employ protocol elements known to
result in efficient operation. Techniques to be considered include:
o Re-use of transport connections across sessions
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o Piggybacking of responses on requests in the reverse direction
o Caching of state across requests
3.5 Invocation of services
The SPEECHSC framework MUST be compliant with the IAB OPES [3]
framework. The applicability of the SPEECHSC protocol will therefore
be specified as occurring between clients and servers at least one of
which is operating directly on behalf of the user requesting the
service.
3.6 Location and Load Balancing
To the extent feasible, the SPEECHSC framework SHOULD exploit
existing schemes for supporting service location and load balancing,
such as the Service Location Protocol [12] or DNS SRV records [13].
Where such facilities are not deemed adequate, the SPEECHSC framework
MAY define additional load balancing techniques.
3.7 Multiple services
The SPEECHSC framework MUST permit multiple services to operate on a
single media stream so that either the same or different servers may
be performing speech recognition, speaker identification or
verification, etc. in parallel.
3.8 Multiple media sessions
The SPEECHSC framework MUST allow a 1:N mapping between session and
RTP channels. For example, a single session may include an outbound
RTP channel for TTS, an inbound for ASR and a different inbound for
SI/SV (e.g. if processed by different elements on the Media Resource
Element). Note: All of these can be described via SDP, so if SDP is
utilized for media channel description, this requirement is met "for
free".
3.9 Users with disabilities
The SPEECHSC framework must have sufficient capabilities to address
the critical needs of people with disabilities. In particular, the
set of requirements set forth in RFC3351 [4] MUST be taken into
account by the framework. It is also important that implementers of
SPEECHSC clients and servers be cognizant that some interaction
modalities of SPEECHSC may be inconvenient, or simply inappropriate
for disabled users. Hearing-impaired individuals may find TTS of
limited utility. Spech-impaired users may be unable to make use of
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ASR or SI/SV capabilities. Therefore, systems employing SPEECHSC MUST
provide alternative interaction modes or avoid the use of speech
processing entirely.
3.10 Identification of process which produced media or control output
The client of a SPEECHSC operation SHOULD be able to ascertain via
the SPEECHSC framework what speech process produced the output. For
example, an RTP stream containing the spoken output of TTS should be
identifiable as TTS output, and the recognized utterance of ASR be
identifiable as having been produced by ASR processing.
4. TTS Requirements
4.1 Requesting Text Playback
The SPEECHSC framework MUST allow a Media Processing Entity or
Application Server, using a control protocol, to request the TTS
Server to playback text as voice in an RTP stream.
4.2 Text Formats
4.2.1 Plain Text
The SPEECHSC framework MAY assume that all TTS servers are capable of
reading plain text. For reading plain text, framework MUST allow the
language and voicing to be indicated via session parameters. For
finer control over such properties, see Section 4.2.2.
4.2.2 SSML
The SPEECHSC framework MUST support SSML[3] <speak> basics, and
SHOULD support other SSML tags. The framework assumes all TTS servers
are capable of reading SSML formatted text. Internationalization of
TTS in the SPEECHSC framework, including multi-lingual output within
a single utterance, is accomplished via SSML xml:lang tags.
4.2.3 Text in Control Channel
The Speechsc framework assumes all TTS servers accept text over the
SPEECHSC connection for reading over the RTP connection. The
framework assumes the server can accept text either "by value"
(embedded in the protocol), or "by reference" (e.g. by de-referencing
a URI embedded in the protocol).
4.2.4 Document Type Indication
A document type specifies to the syntax in which the text to be read
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is encoded. The SPEECHSC framework MUST be capable of explicitly
indicating the document type of the text to be processed, as opposed
to forcing the server to infer the content by other means.
4.3 Control Channel
The SPEECHSC framework MUST be capable of establishing the control
channel between the client and server on a per-session basis, where a
session is loosely defined to be associated with a single "call" or
"dialog". The protocol SHOULD be capable of maintaining a long-lived
control channel for multiple sessions serially, and MAY be capable of
shorter time horizons as well, including as short as for the
processing of a single utterance.
4.4 Media origination/termination by control elements
The SPEECHSC framework MUST NOT require the controlling element
(application server, media processing entity) to accept or originate
media streams. Media streams MAY source & sink from the controlled
element (ASR, TTS, etc.).
4.5 Playback Controls
The Speechsc framework MUST support "VCR controls" for controlling
the playout of streaming media output from SPEECHSC processing, and
MUST allow for servers with varying capabilities to accommodate such
controls. The protocol SHOULD allow clients to state what controls
they wish to use, and for servers to report which ones they honor.
These capabilities include:
o The ability to jump in time to the location of a specific marker.
o The ability to jump in time, forwards or backwards, by a specified
amount of time. Valid time units MUST include seconds, words,
paragraphs, sentences, and markers.
o The ability to increase and decrease playout speed.
o The ability to fast-forward and fast-rewind the audio, where
snippets of audio are played as the server moves forwards or
backwards in time.
o The ability to pause and resume playout.
o The ability to increase and decrease playout volume.
These controls SHOULD be made easily available to users through the
client user interface and through per-user customization capabilities
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of the client. This is particularly important for hearing-impaired
users, who will likely desire settings and control regimes different
from those that would be acceptable for non-impaired users.
4.6 Session Parameters
The SPEECHSC framework MUST support the specification of session
parameters, such as language, prosody and voicing.
4.7 Speech Markers
The SPEECHSC framework MUST accommodate speech markers, with
capability at least as flexible as that provided in SSML [1]. The
framework MUST further provide an efficient mechanism for reporting
that a marker has been reached during playout.
5. ASR Requirements
5.1 Requesting Automatic Speech Recognition
The SPEECHSC framework MUST allow a Media Processing Entity or
Application Server to request the ASR Server to perform automatic
speech recognition on an RTP stream, returning the results over
SPEECHSC.
5.2 XML
The Speechsc framework assumes that all ASR servers support the
VoiceXML speech recognition grammar specification (SRGS) for speech
recognition [2].
5.3 Grammar Requirements
5.3.1 Grammar Specification
The SPEECHSC framework assumes all ASR servers are capable of
accepting grammar specifications either "by value" (embedded in the
protocol), or "by reference" (e.g. by de-referencing a URI embedded
in the protocol). The latter MUST allow the indication of a grammar
already known to, or otherwise "built in" to the server. The
framework and protocol further SHOULD exploit the ability to store
and later retrieve by reference large grammars which were originally
supplied by the client.
5.3.2 Explicit Indication of Grammar Format
The SPEECHSC framework protocol MUST be able to explicitly convey the
grammar format in which the grammar is encoded and MUST be extensible
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to allow for conveying new grammar formats as they are defined.
5.3.3 Grammar Sharing
The SPEECHSC framework SHOULD exploit sharing grammars across
sessions for servers which are capable of doing so. This supports
applications with large grammars for which it is unrealistic to
dynamically load. An example is a city-country grammar for a weather
service.
5.4 Session Parameters
The SPEECHSC framework MUST accommodate at a minimum all of the
protocol parameters currently defined in MRCP [9] In addition there
SHOULD be a capability to reset parameters within a session.
5.5 Input Capture
The SPEECHSC framework MUST support a method directing the ASR Server
to capture the input media stream for later analysis and tuning of
the ASR engine.
6. Speaker Identification and Verification Requirements
6.1 Requesting SI/SV
The SPEECHSC framework MUST allow a Media Processing Entity to
request the SI/SV Server to perform speaker identification or
verification on an RTP stream, returning the results over SPEECHSC.
6.2 Identifiers for SI/SV
The SPEECHSC framework MUST accommodate an identifier for each
verification resource and permit control of that resource by ID,
because voiceprint format and contents are vendor specific.
6.3 State for multiple utterances
The SPEECHSC framework MUST work with SI/SV servers which maintain
state to handle multi-utterance verification.
6.4 Input Capture
The SPEECHSC framework MUST support a method for capturing the input
media stream for later analysis and tuning of the SI/SV engine. The
framework may assume all servers are capable of doing so. In addition
the framework assumes that the captured stream contains enough
timestamp context (e.g. the NTP time range from the RTCP packets
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which corresponds to the RTP timestamps of the captured input) to
ascertain after the fact exactly when the verification was requested.
6.5 SI/SV functional extensibility
The SPEECHSC framework SHOULD be extensible to additional functions
associated with SI/SV, such as prompting, utterance verification, and
retraining.
7. Duplexing and Parallel Operation Requirements
One very important requirement for an interactive speech-driven
system is that user perception of the quality of the interaction
depends strongly on the ability of the user to interrupt a prompt or
rendered TTS with speech. Interrupting, or barging, the speech
output requires more than energy detection from the user's direction.
Many advanced systems halt the media towards the user by employing
the ASR engine to decide if an utterance is likely to be real speech,
as opposed to a cough, for example.
7.1 Full Duplex operation
To achieve low latency between utterance detection and halting of
playback, many implementations combine the speaking and ASR
functions. The SPEECHSC framework MUST support such full-duplex
implementations.
7.2 Multiple services in parallel
Good spoken user interfaces typically depend upon the ease with which
the user can accomplish his or her task. When making use of Speaker
Identification or Verification technologies, user interface
improvements often come from the combination of the different
technologies: simultaneous identity claim and verification (on the
same utterance), simultaneous knowledge and voice verification (using
ASR and verification simultaneously). Using ASR and verification on
the same utterance is in fact the only way to support rolling or
dynamically-generated challenge phrases (e.g., "say 51723"). The
SPEECHSC framework MUST support such parallel service
implementations.
7.3 Combination of services
It is optionally of interest that the SPEECHSC framework support more
complex remote combination and controls of speech engines:
o Combination in series of engines that may then act on the input or
output of ASR, TTS or Speaker recognition engines. The control MAY
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then extend beyond such engines to include other audio input and
output processing and natural language processing.
o Intermediate exchanges and coordination between engines
o Remote specification of flows between engines.
These capabilities MAY benefit from service discovery mechanisms
(e.g. engines, properties and states discovery).
8. Additional Considerations (non-normative)
The framework assumes that SDP will be used to describe media
sessions and streams. The framework further assumes RTP carriage of
media, however since SDP can be used to describe other media
transport schemes (e.g. ATM) these could be used if they provide the
necessary elements (e.g. explicit timestamps).
The working group will not be defining distributed speech recognition
methods (DSR), as exemplified by the ETSI Aurora project. The
working group will not be recreating functionality available in other
protocols, such as SIP or SDP.
TTS looks very much like playing back a file. Extending RTSP looks
promising for when one requires VCR controls or markers in the text
to be spoken. When one does not require VCR controls, SIP in a
framework such as Network Announcements [11] works directly without
modification.
ASR has an entirely different set of characteristics. For barge-in
support, ASR requires real-time return of intermediate results.
Barring the discovery of a good reuse model for an existing protocol,
this will most likely become the focus of SPEECHSC.
9. Security Considerations
Protocols relating to speech processing must take security into
account. Many applications of speech technology deal with sensitive
information, such as the use of TTS to read financial information.
Likewise, popular uses for ASR include executing financial
transactions and shopping.
There are at least three aspects of speech processing security which
intersect with the SPEECHSC requirements - securing the speechsc
protocol itself, implementing and deploying the servers which run the
protocol, and ensuring utilization of the technology for providing
security functions is appropriate. Each of these aspects in discussed
in the following sub-sections. While some of these considerations are
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strictly speaking out of scope of the protocol itself, they will be
carefully considered and accommodated during protocol design, and
will be called out as part of the applicability statement
accompanying the protocol specification(s).
9.1 SPEECHSC protocol security
The SPEECHSC protocol MUST in all cases support authentication,
authorization, and integrity, and SHOULD support confidentiality. For
privacy sensitive applications the protocol MUST support
confidentiality. We envision that rather than providing
protocol-specific security mechanisms in SPEECHSC itself, the
resulting protocol will employ security machinery of either a
containing protocol or the transport on which it runs. For example,
we will consider solutions such as using TLS for securing the control
channel, and SRTP for securing the media channel. Third-party
dependencies necessitating transitive trust will be minimized or
explicitly dealt with through the authentication and authorization
aspects of the protocol design.
9.2 Client and server implementation and deployment
Given the possibly sensitive nature of the information carried,
SPEECHSC clients and servers need to take steps to ensure
confidentiality and integrity of the data and its transformations to
and from spoken form. In addition to these general considerations,
certain SPEECHSC functions, such as speaker verification and
identification, employ voiceprints whose privacy and integrity must
be maintained. Similarly, the requirement to support input capture
for analysis and tuning can represent a privacy vulnerability because
user utterances are recorded and could be either revealed or
re-played inappropriately.
9.3 Use of SPEECHSC for security functions
Speaker identification/verification can be used as an authentication
and/or authorization technology. When so employed, there are
additional vulnerabilties that may need to be addressed through the
use of security measures for both the protocol and the clients and
servers. For example, the ability of manipulate the media stream of a
speaker verification request could inappropriately permit or deny
access based on impersonation, or simple garbling via noise
injection. Further discussion of the use of SI/SV as an
authentication technology, and some recommendations concerning
advantages and vulnerabilities, see [14]
10. Acknowledgements
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Eric Burger wrote the original draft of these requirements and has
continued to contribute actively throughout their development. He is
a co-author in all but formal authorship, and is instead acknowledged
here as it is preferable that working group co-chairs have
non-conflicting roles with respect to the progression of documents.
Normative References
[1] World Wide Web Consortium, "Speech Synthesis Markup Language
Version 1.0", W3C Working Draft , December 2002, <http://
www.w3.org/TR/2002/WD-speech-synthesis-20021202/ >.
[2] World Wide Web Consortium, "Speech Recognition Grammar
Specification Version 1.0", W3C Candidate Recommendation , June
2002, <http://www.w3.org/TR/2002/CR-speech-grammar-20020626/>.
[3] Floyd, S. and L. Daigle, "IAB Architectural and Policy
Considerations for Open Pluggable Edge Services", RFC 3238,
January 2002.
[4] Charlton, N., Gasson, M., Gybels, G., Spanner, M. and A. van
Wijk, "User Requirements for the Session Initiation Protocol
(SIP) in Support of Deaf, Hard of Hearing and Speech-impaired
Individuals", RFC 3351, August 2002.
Informative References
[5] 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.
[6] Arango, M., Dugan, A., Elliott, I., Huitema, C. and S. Pickett,
"Media Gateway Control Protocol (MGCP) Version 1.0", RFC 2705,
October 1999.
[7] Cuervo, F., Greene, N., Rayhan, A., Huitema, C., Rosen, B. and
J. Segers, "Megaco Protocol Version 1.0", RFC 3015, November
2000.
[8] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
[9] Shanmugham, S., Monaco, P. and B. Eberman, "MRCP: Media
Resource Control Protocol", Internet Draft
draft-shanmugham-mrcp-04.txt, May 2003.
[10] World Wide Web Consortium, "Voice Extensible Markup Language
(VoiceXML) Version 2.0", W3C Working Draft , April 2002,
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<http://www.w3.org/TR/2002/WD-voicexml20-20020424/>.
[11] Burger, E., Van Dyke, J., O'Connor, W. and A. Spitzer, "Basic
Network Media Services with SIP",
draft-burger-sipping-netann-05 (work in progress), March 2003.
[12] Guttman, E., Perkins, C., Veizades, J. and M. Day, "Service
Location Protocol, Version 2", RFC 2608, June 1999.
[13] Gulbrandsen, A., Vixie, P. and L. Esibov, "A DNS RR for
specifying the location of services (DNS SRV)", RFC 2782,
February 2000.
[14] Committee on Authentication Technologies and Their Privacy
Implications, National Research Council, "Who Goes There?:
Authentication Through the Lens of Privacy", Computer Science
and Telecommunications Board (CSTB) , 2003, <http://
www.nap.edu/catalog/10656.html/ >.
Author's Address
David R Oran
Cisco Systems, Inc.
7 Ladyslipper Lane
Acton, MA
USA
EMail: oran@cisco.com
Oran Expires December 5, 2003 [Page 16]
Internet-Draft Speech Services Control Requirements June 2003
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Internet-Draft Speech Services Control Requirements June 2003
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Oran Expires December 5, 2003 [Page 18]
