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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   http://www.ietf.org/ietf/1id-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at
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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
   subscribe, send the message "subscribe mrcp" to
   majordomo@snowshore.com.  The public archive is at
   http://flyingfox.snowshore.com/mrcp_archive/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].


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   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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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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   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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   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 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

   12 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

   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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