Network Working Group E. Burger
Internet Draft SnowShore Networks, Inc.
Document: draft-burger-mrcp-reqts-00.txt D. Oran
Category: Informational Cisco Systems, Inc.
Expires August 2002 February 19, 2002
Requirements for Distributed Control of ASR 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].
Internet-Drafts are working documents of the Internet Engineering
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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 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 MRCP list. To
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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].
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Distributed Media Control Requirements February 2002
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]). The focus of
this document is requirements for one or more protocols to support
the control of network elements that perform Automated Speech
Recognition (ASR) 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 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 whenever possible.
5.2. Maintain Existing Protocol Integrity
In meeting requirement 5.1, the SRCP framework MUST NOT redefine the
semantics of an existing protocol.
Said differently, we will not break existing protocols.
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Distributed Media Control Requirements February 2002
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.
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 [11] 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.
OPEN ISSUE: Should we allow the TTS Server to retrieve text on its
own? That is, have SRCP pass in a URI from which the TTS Server
retrieves the text.
OPEN ISSUE: Should we allow (or require) the TTS Server to use long-
lived control channels?
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.
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The ASR Server MUST support the XML specification for speech
recognition [12].
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. 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.
9. 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 [13] works directly without
modification.
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Distributed Media Control Requirements February 2002
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.
10. 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 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.
11. 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
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Distributed Media Control Requirements February 2002
9 World Wide Web Consortium, "Voice Extensible Markup Language
(VoiceXML) Version 2.0", W3C Working Draft,
,
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 World Wide Web Consortium, "Speech Synthesis Markup Language
Specification for the Speech Interface Framework", W3C Working
Draft, , January 2001,
work in progress
12 World Wide Web Consortium, "Speech Recognition Grammar
Specification for the W3C Speech Interface Framework", W3C
Working Draft, , August
2001, work in progress
13 O'Connor, W., Burger, E., "Network Announcements with SIP",
draft-ietf-sipping-netann-01.txt, November 2001, work in progress
12. Acknowledgments
Brian Eberman came up with the new name. It is catchy and describes
what we are working on.
OPEN ISSUE: Chose a name!
13. 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
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Distributed Media Control Requirements February 2002
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