Network Working Group E. Burger
Internet Draft SnowShore Networks, Inc.
Document: draft-ietf-speechsc-reqts-00.txt D. Oran
Category: Informational Cisco Systems, Inc.
Expires January 2003 July 31, 2002
Requirements for Distributed Control of ASR, SR and TTS Resources
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026 [1].
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1. 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, speaker recognition (which includes both speaker
identification and speaker verification) and text-to-speech. 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.
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. Conventions used in this document
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 RFC-2119 [2].
FORMATTING NOTE: Notes, such at this one, provide additional,
nonessential information that the reader may skip without missing
anything essential. The primary purpose of these non-essential
notes is to convey information about the rationale of this document,
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or to place this document in the proper historical or evolutionary
context. Readers whose sole purpose is to construct a conformant
implementation may skip such information. However, it may be of use
to those who wish to understand why we made certain design choices.
OPEN ISSUES: This document highlights questions that are, as yet,
undecided as "OPEN ISSUES".
3. Introduction
There are multiple IETF protocols for establishment and termination
of media sessions (SIP[3]), low-level media control (MGCP[4] and
MEGACO[5]), and media record and playback (RTSP[6]). This document
focuses on requirements for one or more protocols to support the
control of network elements that perform Automated Speech
Recognition (ASR), speaker recognition (SR), and rendering text into
audio, a.k.a. 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 on the
distributed control of ASR, SR and TTS servers.
To date, there are a number of proprietary ASR and TTS API's, as
well as two IETF drafts that address this problem [7] [8]. 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 "new protocol" that
addresses the distributed control of speech resources It refers to
such a protocol as "SPEECHSC", for Speech Services Control Protocol.
4. SPEECHSC Framework
The following is the SPEECHSC framework for speech processing.
+-------------+
| Application |
| Server |\
+-------------+ \ SPEECHSC
SIP or whatever / \
/ \
+------------+ / \ +--------+
| Media |/ SPEECHSC \---| ASR |
| Processing |-------------------------| and/or |
RTP | Entity | RTP | TTS |
=====| |=========================| Server |
+------------+ +--------+
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The "Media Processing Entity" is a network element that processes
media. The "Application Server" is a network element that instructs
the Media Processing Entity on what transformations to make to the
media stream. The "ASR and/or TTS Server" is a network element that
either generates a RTP stream based on text input (TTS) or returns
speech recognition results in response to an RTP stream as input
(ASR). Either the Media Processing Entity or the Application Server
may control the ASR or TTS Server using SPEECHSC as a control
protocol.
Physical embodiments of the entities can reside in one physical
instance per entity, or some combination of entities. For example,
a VoiceXML [9] 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.
5. General Requirements
5.1. Reuse Existing Protocols
To the extent feasible, the SPEECHSC framework SHOULD use existing
protocols.
5.2. Maintain Existing Protocol Integrity
In meeting requirement 5.1, 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.
5.3. Avoid Duplicating Existing Protocols
To the extent feasible, SPEECHSC SHOULD NOT duplicate the
functionality of existing protocols. For example, SIP with msuri
[10] and RTSP 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
requirement 5.2. Where an existing protocol can be gracefully
extended to support SPEECHSC requirements, such extensions are
acceptable alternatives for meeting the requirements.
5.4. Explicit invocation of services
The SPEECHSC framework MUST be compliant with the IAB OPES[11]
framework. The applicability of the SPEECHSC protocol will therefore
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be specified as occurring between clients and servers at least one
of which is operating directly on behalf of the user requesting the
service.
5.5. Server Location and Load Balancing
To the extent feasible, the SPEECHSC framework SHOULD exploit
existing schemes for performing 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.
5.6. Simultaneous 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 recognition, etc. in
parallel.
6. TTS Requirements
The SPEECHSC framework MUST allow a Media Processing Entity, using a
control protocol, to request the TTS Server to playback text as
voice in an RTP stream.
The TTS Server MUST support the reading of plain text. For reading
plain text, the language and voicing MAY be indicated via session
parameters. For finer control over such properties, use of SSML
rather than plain text provides the necessary capabilities.
The TTS Server SHOULD support the reading of SSML [14] text.
The TTS Server MUST accept text over the SPEECHSC connection for
reading over the RTP connection. The server MUST accept text either
?by value? (embedded in the protocol), or ?by reference? (by de-
referencing a URI embedded in the protocol).
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.
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.
The TTS Server SHOULD support, and the SPEECHSC framework MUST
support the specification of, "VCR Controls":
1. The ability to jump in time to the location of a specific
marker.
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2. 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.
3. The ability to increase and decrease playout speed.
4. 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.
5. The ability to pause and resume playout.
6. The ability to increase and decrease playout volume.
The SPEECHSC framework must support the specification of session
parameters, such as language, prosody and voicing.
The SPEECHSC framework MUST accommodate speech markers, with
capability at least as flexible as that provided in SSML[14]. The
framework MUST further provide an efficient mechanism for reporting
that a marker has been reached during playout.
7. ASR Requirements
The SPEECHSC framework MUST allow a Media Processing Entity to
request the ASR Server to perform automatic speech recognition on an
RTP stream, returning the results over SPEECHSC.
The ASR Server MUST support the XML specification for speech
recognition [15].
The ASR Server MUST accept grammar specifications either ?by value?
(embedded in the protocol), or ?by reference? (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. Servers SHOULD be able to store and later retrieve by
reference large grammars which were originally supplied by the
client.
The SPEECHSC framework protocol MUST be able to explicitly convey
the grammar format in which the grammar is encoded and MUST be
extensible to allow for conveying new grammar formats as they are
defined.
The SPEECHSC framework MUST accommodate at a minimum all of the
protocol parameters currently defined in MRCP[7]. In addition there
SHOULD be a capability to reset parameters within a session.
The SPEECHSC framework SHOULD support a method directing the ASR
Server to capture the input media stream for later analysis and
tuning of the ASR engine.
The ASR Server SHOULD support sharing grammars across sessions.
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.
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8. Speaker Recognition Requirements
The SPEECHSC framework MUST allow a Media Processing Entity to
request the SR Server to perform speaker recognition on an RTP
stream, returning the results over SPEECHSC.
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
The SPEECHSC framework MUST work with SR servers which maintain
state to handle multi-utterance verification.
The SPEECHSC framework, and SR Server SHOULD support a method for
capturing the input media stream for later analysis and tuning of
the SR engine. Further, the SPEECHSC framework SHOULD be extensible
to associated functions, such as prompting, utterance verification,
and retraining.
9. Dual-Mode 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.
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 dual-mode
implementations.
Good spoken user interfaces typically depend upon the ease with
which the user can accomplish his or her task. When making use of
Speaker Recognition 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 dual-mode implementations.
10. Thoughts to Date (non-normative)
The protocol assumes RTP carriage of media. Assuming session-
oriented media transport, the protocol will use SDP to describe the
session.
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The working group will not be investigating distributed speech
recognition (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 [10] 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.
11. Security Considerations
Protocols relating to speech processing must take security into
account. This is particularly important as popular uses for TTS
include reading financial information. Likewise, popular uses for
ASR include executing financial transactions and shopping.
We envision that rather than providing application-specific security
mechanisms in SPEECHSC itself, the resulting protocol will employ
security machinery of either containing protocols 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-part dependencies necessitating transitive
trust will be minimized or explicitly dealt with through the
authentication and authorization aspects of the protocol design.
In addition to the security machinery needed by the protocol itself,
there are considerations for the implementation and deployment of
the clients and servers themselves. For example, speaker verifica-
tion and identification employs voiceprints whose privacy and
integrity must be maintained. While strictly speaking out of scope
of the protocol itself, such considerations 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).
12. References
1 Bradner, S., "The Internet Standards Process -- Revision 3", BCP
9, RFC 2026, October 1996.
2 Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997
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Distributed Media Control Requirements February 2002
3 Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, H., Schooler, E., "SIP: Session
Initiation Protocol", RFC 3261, June 2002.q
4 Arango, M., Dugan, A., Elliott, I., Huitema, C., and Pickett, S.,
"Media Gateway Control Protocol (MGCP) Version 1.0", RFC 2705,
October 1999
5 Cuervo, F., Greene, N., Rayhan, A., Huitema, C., Rosen, B., and
Segers, J., "Megaco Protocol Version 1.0", RFC 3015, November 2000
6 Schulzrinne, H., Rao, A., and Lanphier, R., "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998
7 Shanmugham, S., Monaco, P., and B. Eberman, "MRCP: Media Resource
Control Protocol", draft-shanmugham-mrcp-02.txt, July 2002, work
in progress
8 Robinson, F., Marquette, B., and R. Hernandez, "Using Media
Resource Control Protocol with SIP", draft-robinson-mrcp-sip-
00.txt, January 2002, work in progress
9 World Wide Web Consortium, "Voice Extensible Markup Language
(VoiceXML) Version 2.0", W3C Working Draft,
,
April 2002, work in progress
10 Van Dyke, J., Burger, E., Spitzer, A., O'Connor, W., "Basic
Network Media Services with SIP", draft-burger-sipping-netann-
02.txt, June 2002, work in progress
11 Floyd, S., Daigle, L., ?IAB Architectural and Policy
Considerations for Open Pluggable Edge Services,? RFC3238,
January 2002.
12 Guttman, E., Perkins, C., Veizades, J., Day, M. , "Service
Location Protocol, Version 2,? RFC 2608, June 1999.
13 Gulbrandson, A, Vixie, P., Esibov, L., ?A DNS RR for specifying
the location of services (DNS SRV)?, RFC2782, February 2000.
14 World Wide Web Consortium, "Speech Synthesis Markup Language
Specification for the Speech Interface Framework", W3C Working
Draft 5, ,
April 2002, work in progress
15 World Wide Web Consortium, "Speech Recognition Grammar
Specification Version 1.0", W3C Candidate Recommendation,
, June
2002, work in progress
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Distributed Media Control Requirements February 2002
13. Acknowledgments
Brian Eberman came up with the new name. It is catchy and describes
what we are working on.
14. Author's Addresses
Eric W. Burger
SnowShore Networks, Inc.
Chelmsford, MA
USA
Email: eburger@snowshore.com
David R. Oran
Cisco Systems, Inc.
Acton, MA
USA
Email: oran@cisco.com
15. Change Log
From version draft-burger-mrcp-reqts-00 to version draft-burger-
speechsc-reqts-00:
- draft name changed per area director advice
- added speaker verification to the areas addressed, including
speaker verification requirements, per Dan Burnet?s
presentation at the Minneapolis BoF (see minutes).
- based on mailing list discussion, added requirement to handle
both ?by value? and ?by reference? data. This is both for TTS
to be played out and grammar(s) to be applied to ASR.
- Based on discussion at the BoF in Minneapolis, added a
requirement concerning the use of load balancing schemes,
including those based on SRVLOC, SRV.
- Added a requirement for OPES compliance, per a discussion
with Sally Floyd as IAB observer for the BoF.
From version draft-burger-speechsc-reqts-00 to version draft-ietf-
speechsc-reqts-00:
- Changed ?SV? to ?SR? and ?speaker verification? to ?speaker
recognition? everywhere
- Replaced SRCP with SPEECHSC everywhere
- Minor edits including mailing list name change, temporary
notes removed,
- All agreements reached at the IETF 54 WG meeting, confirmed
by mailing list discussion, up through 8/10/02 have been
integrated
- Improved requirement on VCR controls as suggested by Dan
Burnett and Sarvi Shanmughan
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- Text describing dual-mode requirements for ASR and SR by Dan
Burnett added.
- Suggested change to framework figure made by Rajiv
Dharmadhikari incorporated
- Updated references to most recent versions
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