Internet DRAFT - draft-burger-speechsc-reqts
draft-burger-speechsc-reqts
Network Working Group E. Burger
Internet Draft SnowShore Networks, Inc.
Document: draft-burger-speechsc-reqts-00.txt D. Oran
Category: Informational Cisco Systems, Inc.
Expires August 2002 June 13, 2002
Requirements for Distributed Control of ASR, SV 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 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.
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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,
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 verification (SV), 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, SV 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 "SRCP", for Speech
Resource Control Protocol.
4. SRCP Framework
The following is the SRCP framework for speech processing.
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+-------------+
| Application |
| Server |
+-------------+
SIP or whatever /
/
+------------+ / +--------+
| Media |/ SRCP | ASR |
| Processing |-------------------------| and/or |
RTP | Entity | RTP | TTS |
=====| |=========================| Server |
+------------+ +--------+
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). The Media Processing Entity controls the ASR or TTS Server
using SRCP 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 SRCP.
5. General Requirements
5.1. Reuse Existing Protocols
To the extent feasible, the SRCP framework SHOULD use existing
protocols.
5.2. Maintain Existing Protocol Integrity
In meeting requirement 5.1, the SRCP framework MUST NOT redefine the
semantics of an existing protocol.
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Said differently, we will not break existing protocols or cause
backward compatibility problems.
5.3. Avoid Duplicating Existing Protocols
To the extent feasible, SRCP 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 SRCP is new functionality not addressed by existing
protocols or extending existing protocols within the strictures of
requirement 5.2.
5.4. Explicit invocation of services
The SRCP framework MUST be compliant with the IAB OPES[11]
framework. The applicability of the SRCP 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.
5.5. Server Location and Load Balancing
To the extent feasible, the SRCP 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 SRCP framework MAY define
additional load balancing techniques.
6. TTS Requirements
The SRCP 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 is a local matter.
The TTS Server SHOULD support the reading of SSML [14] text.
OPEN ISSUE: Should the TTS Server infer the text is SSML by
detecting a legal SSML document, or must the protocol tell the TTS
Server the document type?
The TTS Server MUST accept text over the SRCP 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).
OPEN ISSUE: Should we allow (or require) the TTS Server to use long-
lived control channels?
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The TTS Server SHOULD support, and the SRCP framework MUST support
the specification of, "VCR Controls", such as skip forward, skip
backward, play faster, and play slower.
OPEN ISSUE: Should we allow for session parameters, like prosody and
voicing, as is specified for MRCP over RTSP [7]?
OPEN ISSUE: Should we allow for speech markers, as is specified for
MRCP over RTSP [7]?
7. ASR Requirements
The SRCP 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 SRCP.
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).
OPEN ISSUE: Should we allow the ASR Server to support alternative
grammar formats? If so, we need mechanisms to specify what format
the grammar is in, capability discovery, and handling unsupported
grammars.
OPEN ISSUE: Is there a need for all of the parameters specified for
MRCP over RTSP [7]? Most of them are part of the W3C speech
recognition grammar.
The ASR Server SHOULD support a method for capturing 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.
8. Speaker Verification Requirements
The SRCP framework MUST allow a Media Processing Entity to request
the SV Server to perform speaker verification on an RTP stream,
returning the results over SRCP.
The SV Server MUST The 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 SRCP 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
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The SRCP framework MUST work with SV servers which maintain state to
handle multi-utterance verification.
The SV Server SHOULD support a method for capturing the input media
stream for later analysis and tuning of the SV engine.
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 SRCP 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.
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 [16] 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 SRCP.
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.
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We envision that rather than providing application-specific security
mechanisms in SRCP 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.
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
3 Handley, M., Schulzrinne, H., Schooler, E., and Rosenberg, J.,
"SIP: Session Initiation Protocol", RFC 2543, March 1999
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-01.txt, November 2001,
work in progress
8 Robinson, F., Marquette, B., and R. Hernandez, "Using Media
Resource Control Protocol with SIP", draft-robinson-mrcp-sip-
00.txt, September 2001, work in progress
9 World Wide Web Consortium, "Voice Extensible Markup Language
(VoiceXML) Version 2.0", W3C Working Draft,
<http://www.w3.org/TR/2001/WD-voicexml20-20011023/>,
October 2001, work in progress
10 Van Dyke, J. and Burger, E., "SIP URI Conventions for Media
Servers", draft-burger-sipping-msuri-01, July 2001, work in
progress (expired)
11 Floyd, S., Daigle, L., ?IAB Architectural and Policy
Considerations for Open Pluggable Edge Services,? RFC3238,
January 2002.
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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, <http://www.w3.org/TR/speech-synthesis>, January 2001,
work in progress
15 World Wide Web Consortium, "Speech Recognition Grammar
Specification for the W3C Speech Interface Framework", W3C
Working Draft, <http://www.w3.org/TR/speech-grammar/>, August
2001, work in progress
16 O'Connor, W., Burger, E., "Network Announcements with SIP",
draft-ietf-sipping-netann-01.txt, November 2001, work in progress
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).
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- 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.
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