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From: "Brian Wyld" <brian.wyld@eloquant.com>
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Cc: <eburger@snowshore.com>, "'Speechsc (E-mail)'" <speechsc@ietf.org>,
        <brian.wyld@eloquant.com>
Date: Fri, 1 Nov 2002 21:01:14 +0100
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Subject: [Speechsc] speechsc WG Internet Draft submission : draft-ietf-speechsc-protocol-eval-01.txt  - Update to version 01 for  protocol evaluation document
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Hi,

Attached is an update of the speechsc protocol evaluation draft for
submission as Internet draft.

Thanks for processing this update.

A+

Brian

[Brian Wyld] [brian.wyld@eloquant.com]
[Directeur General R&D]
[Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
[advanced solutions for telecoms and IT services]


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=0A=
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 Internet Draft                                               B. Wyld=20
 Document: draft-speechsc-protocol-eval.txt                     Editor=20
 Expires: April 2003                                          Eloquant=20
 Version 01                                             November 2002=20
 =20
                       SPEECHSC Protocol Evaluation=20
     =20
 Status of this Memo =20
    =20
    This document is an Internet-Draft and is in full conformance with=20
    all provisions of Section 10 of RFC2026. =20
        =20
    Internet-Drafts are working documents of the Internet Engineering=20
    Task Force (IETF), its areas, and its working groups.  Note that=20
    other groups may also distribute working documents as Internet-
    Drafts. =20
        =20
    Internet-Drafts are draft documents valid for a maximum of six=20
    months and may be updated, replaced, or obsoleted by other=20
    documents at any time. It is inappropriate to use Internet-Drafts=20
    as reference material or to cite them other than as "work in=20
    progress." =20
        =20
    The list of current Internet-Drafts can be accessed at =20
         http://www.ietf.org/ietf/1id-abstracts.txt =20
    The list of Internet-Draft Shadow Directories can be accessed at =20
         http://www.ietf.org/shadow.html. =20
        =20
 Abstract =20
    =20
    This document is the Protocol Evaluation Document for the SPEECHSC=20
    Working Group.  Section 3 provides the summary of the  individual=20
    protocol comparisons (in the sections 4-N following) against the=20
    SPEECHSC requirements [1].=20
  =20
 Table of Contents=20
    =20
    1.   Overview...................................................2=20
    2.   Protocol Proposals.........................................2=20
    3.   Protocol Evaluation Summaries..............................3=20
       3.1.   Protocol X............................................3=20
    4.   Protocol =93Beep=94 Complience Evaluation (Jerry =
Carter).......4=20
       4.1.   General notes:........................................4=20
       4.2.   Analysis of General Requirements......................4=20
       4.3.   Analysis of TTS requirements..........................5=20
       4.4.   Analysis of ASR requirements..........................6=20
       4.5.   Analysis of Speaker Identification and Verification=20
       Requirements..................................................7=20
       4.6.   Analysis of Duplexing and Parallel Operation=20
       Requirements..................................................7=20
       4.7.   Analysis of additional considerations (non-normative).7=20
       4.8.   Analysis of Security considerations...................8=20
    5.   Protocol =93SIP=94 Complience Evaluation (Rajiv =
Dharmadhikari).8=20
       5.1.   Introduction..........................................8=20
       5.2.   Analysis of General Requirements......................8=20
       5.3.   Analysis of TTS requirements..........................9=20
       5.4.   Analysis of ASR requirements.........................10=20
       5.5.   Analysis of Speaker Identification and Verification=20
       Requirements.................................................10=20
       5.6.   Analysis of Duplexing and Parallel Operation=20
       Requirements.................................................11=20
 =20
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 =20
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       5.7.   Analysis of additional considerations (non-normative)11=20
       5.8.   Analysis of Security considerations..................11=20
       5.9.   Other Criteria.......................................11=20
    6.   Protocol =93RTSP=94 Complience Evaluation (Brian =
Wyld)........12=20
       6.1.   General Introduction.................................12=20
       6.2.   Analysis of General Requirements.....................13=20
       6.3.   Analysis of TTS requirements.........................13=20
       6.4.   Analysis of ASR requirements.........................14=20
       6.5.   Analysis of Speaker Identification and Verification=20
       Requirements.................................................14=20
       6.6.   Analysis of Duplexing and Parallel Operation=20
       Requirements.................................................15=20
       6.7.   Analysis of additional considerations (non-normative)15=20
       6.8.   Analysis of Security considerations..................15=20
    7.   Protocol =93MRCP=94 Complience Evaluation (Sarvi =
Shanmugham)..15=20
       7.1.   General..............................................15=20
       7.2.   Analysis of General Requirements.....................16=20
       7.3.   Analysis of TTS requirements.........................17=20
       7.4.   Analysis of ASR requirements.........................18=20
       7.5.   Analysis of Speaker Identification and Verification=20
       Requirements.................................................19=20
       7.6.   Analysis of Duplexing and Parallel Operation=20
       Requirements.................................................19=20
       7.7.   Analysis of additional considerations (non-normative)20=20
       7.8.   Analysis of Security considerations..................20=20
    8.   Protocol =93Web Services=94 Complience Evaluation (Stephane H.=20
    Maes) =0D         20=20
       8.1.   General Notes:.......................................20=20
       8.2.   Analysis of General Requirements.....................21=20
       8.3.   Analysis of TTS requirements.........................23=20
       8.4.   Analysis of ASR requirements.........................24=20
       8.5.   Analysis of Speaker Identification and Verification=20
       Requirements.................................................25=20
       8.6.   Analysis of Duplexing and Parallel Operation=20
       Requirements.................................................25=20
       8.7.   Analysis of additional considerations (non-normative)26=20
       8.8.   Analysis of Security considerations..................26=20
    9.   Security Considerations...................................27=20
    10.  References................................................27=20
       =20
      1. =0D         Overview =20
        =20
    This document provides the template for the content for the=20
    SPEECHSC Protocol Evaluation document.   =20
        =20
    This section will contain an overview of the process.=20
    =20
    Section 2 contains a list of the proposed protocols submitted to=20
    WG.=20
    =20
    Section 3 provides a summary of the proposed protocols against the=20
    Requirements and framework. =20
    =20
    Sections 4 and following provide the individual protocol=20
    comparisons against the SPEECHSC requirements [1].    =20
        =20
      2. =0D         Protocol Proposals=20
 =20
    This section contains a list of the existing protocols submitted to=20
    the SPEECHSC WG for consideration by the deadline. =20
 =20
 =20
 Wyld                    Expires =96 April 2003               [Page 2] =
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                 SPEECHSC Protocol Evaluation Template  November 2002=20
 =20
 =20
          1. BEEP=20
          2. SIP=20
          3. RTSP=20
          4. MRCP (initial submission)=20
          5. Web Services=20
    =20
    =20
    Each protocol section contains a review of the protocol=92s level of =

    compliance to each of the SPEECHSC Requirements [1] as derived from=20
    the proposed protocol documents. The following key will be used to=20
    identify the level of compliancy of each of the individual=20
    protocols:=20
    =20
    T =3D Total Complience.  Meets the requirement fully.=20
    P =3D Partial Compliance.  Meets some aspect of the requirement.=20
    P+ =3D Complience possible. Could meet the requirement with =
=93natural=94=20
    evolution of the protocol. =20
    F =3D Failed Compliance.  Does not meet the requirement. =20
    =20
      3. =0D         Protocol Evaluation Summaries =20
        =20
    To provide standalone completeness for this document, this section=20
    contains a summary of each of the protocol comparisons. The=20
    comparisons should explicitly reference the requirements as defined=20
    in [1]. =20
    =20
    TBD : This section will be completed once agreement is reached on=20
    the specific protocol analysis sections.=20
=0A=
         3.1. Protocol X=20
=0A=
           3.1.1. Protocol x Architectural Model as compared to the=20
              SPEECHSC Architectural Framework=20
    =20
    This section would contain a description of the key aspects of the=20
    architectural model of protocol X and a comparison of this model=20
    against the SPEECHSC framework, highlighting the pros/cons of the=20
    applicability of protocol x to SPEECHSC.=20
=0A=
           3.1.2. SPEECHSC requirements met by protocol X.=20
    =20
    This section contains a description of the SPEECHSC requirements=20
    met by protocol X.=20
=0A=
           3.1.3. SPEECHSC requirements partially met by protocol X.=20
    =20
    This section contains a description of the SPEECHSC requirements=20
    partially met by protocol X. [Note: ideally, there would be few or=20
    NONE of these provided the SPEECHSC requirements have been defined=20
    at the appropriate level]. =20
=0A=
           3.1.4.   SPEECHSC requirements can be met by protocol X=20
              with =93natural=94 evolutions.=20
    =20
    This section contains a description of the SPEECHSC requirements=20
    not currently met by protocol X but that could be met assuming some=20
    evolutions. [Note: the definition of =93natural=94 evolution will no =

    doubt be for discussion]. =20
 =20
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                 SPEECHSC Protocol Evaluation Template  November 2002=20
 =20
 =20
           3.1.5. SPEECHSC requirements NOT met by protocol X. =20
    =20
    This section contains a description of the SPEECHSC requirements=20
    that are NOT met by protocol X. [Note: an important aspect to=20
    highlight in the individual protocol comparisons would be the work=20
    required to extend protocol X to be able to support this=20
    requirement.]=20
    =20
      4. =0D         Protocol =93Beep=94 Complience Evaluation (Jerry =
Carter)=20
=0A=
         4.1. General notes:=20
    =20
    The BEEP protocol provides a general framework for establishing =20
    connections, defining new channels, negotiating security, and=20
    performing user authentication. Protocols build on beep must define=20
    a profile detailing how connections are established and must define=20
    a set of messages which will be delivered using BEEP. The protocol=20
    is peer-to-peer although client-server style requests could be=20
    easily handled.=20
    =20
    The following sub-sections compare each individual requirement=20
    against the protocol.=20
=0A=
         4.2. Analysis of General Requirements=20
=0A=
           4.2.1. Reuse existing protocols [5.1]=20
    =20
    T: Beep is a published protocol, listed as RFC 3080=20
    (http://www.ietf.org/rfc/rfc3080.txt).=20
=0A=
           4.2.2. Maintain Existing Protocol Integrity [5.2]=20
    =20
    P: BEEP assumes that protocols, such as SpeechSC, will add=20
    messages.=20
    =20
    Supporting multiple clients using TCP may require some effort.=20
=0A=
           4.2.3. Avoid Duplicating Existing Protocols [5.3]=20
    =20
    T: Building SpeechSC over BEEP would allow the specification to=20
    focus on managing the ASR, media server, and SI/SV resources and=20
    the possible interactions between them. The operations for=20
    establishing connections and defining new channels would be handled=20
    by BEEP.=20
=0A=
           4.2.4. Protocol efficiency [5.4]=20
    =20
    P+: BEEP imposes a small overhead (roughly 40 bytes per message).=20
    It provides a mechanism for supporting multiple communication=20
    channels over a single port. If grouping of requests is desired,=20
    this would need to be handled by grouping the SpeechSC messages.=20
=0A=
           4.2.5. Explicit invocation of services [5.5]=20
    =20
    T: Though it is primarily a peer-to-peer protocol, BEEP may act as=20
    a traditional client server protocol.=20
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                 SPEECHSC Protocol Evaluation Template  November 2002=20
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           4.2.6. Server Location and Load Balancing [5.6]=20
    =20
    P+: This functionality is not provided by BEEP. This would need to=20
    be added as an extension.=20
=0A=
           4.2.7. Simultaneous services [5.7]=20
    =20
    T: Multiple channels providing different services is possible. Each=20
    service is simply a message type which is passed to the server=20
    using BEEP.=20
=0A=
           4.2.8. Multiple media sessions [5.8]=20
    =20
    F: BEEP assumes a 1:1 using TCP/IP.=20
=0A=
         4.3. Analysis of TTS requirements=20
=0A=
           4.3.1. Requesting Text Playback [6.1]=20
    P+: A medial playback resource would be defined as a new BEEP=20
    profile.=20
    To initiate an media playback session, the client would need to ask=20
    for this profile as part of the <start> message which opens a new=20
    channel.=20
=0A=
           4.3.2. Text Formats [6.2]=20
    =20
    T: BEEP allows arbitrary data blocks of octets to be passed.=20
    Messages must specify a length and must end with a specific=20
    sequence of characters. Plain text, SSML, URIs, or other content=20
    could be passed within BEEP messages.=20
=0A=
           4.3.3. Plain text [6.2.1]=20
    =20
    T: See <Text Formats>=20
=0A=
           4.3.4. SSML [6.2.2]=20
    =20
    T: See <Text Formats>=20
=0A=
           4.3.5. Text in Control Channel [6.2.3]=20
    =20
    T: See <Text Formats>=20
=0A=
           4.3.6. Document Type Indication [6.2.4]=20
    =20
    T: BEEP provides for 'Content-Type' specification in the message=20
    header. This may be used to determine the media type of the=20
    message.=20
=0A=
           4.3.7. Control Channel [6.3]=20
    =20
    T: A second channel might be created to pass control information.=20
    The BEEP protocol dictates that the requests for each channel must=20
    be handled in the order in which they are received. Separate=20
    channels operate independently and may service request with=20
    different priorities (though the BEEP specification does not=20
    provide a mechanism for assigning priorities).=20
 =20
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                 SPEECHSC Protocol Evaluation Template  November 2002=20
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           4.3.8. Playback Controls [6.4]=20
    =20
    T: See 6.3. These messages would presumably be transmitted over the=20
    control channel.=20
=0A=
           4.3.9. Session Parameters [6.5]=20
    =20
    T: See 6.2. Session parameters are presumably content delivered=20
    within a message.=20
=0A=
           4.3.10. =0D                  Speech Markers [6.6]=20
    =20
    T: Speech markers might be delivered on a separate channel (as in=20
    6.3). =20
    Alternatively, BEEP is a peer-to-peer protocol so events might be=20
    sent on the channel used to receive requests.=20
=0A=
         4.4. Analysis of ASR requirements=20
=0A=
           4.4.1. Requesting Automatic Speech Recognition [7.1]=20
    =20
    P+: An ASR resource would be defined as a new BEEP profile. To=20
    initiate an ASR session, the client would need to ask for this=20
    profile as part of the <start> message which opens a new channel.=20
=0A=
           4.4.2. XML [7.2]=20
    =20
    T: See response for 6.2.=20
=0A=
           4.4.3. Grammar Specification [7.3.1]=20
    =20
    T: See response for 6.2.=20
=0A=
           4.4.4. Explicit Indication of Grammar Format [7.3.2]=20
    =20
    T: See response for 6.2.4.=20
=0A=
           4.4.5. Grammar Sharing [7.3.3]=20
    =20
    P+: Channels in BEEP are independent. If content is to be shared,=20
    this would need to be added, preferably using messages to define=20
    what content is to be shared and a corresponding identity. The=20
    retrieval and actual sharing would need to be implemented by the=20
    server.=20
=0A=
           4.4.6. Session Parameters [7.4]=20
    =20
    T: See response for 6.2. Session parameters are presumably content=20
    delivered within a message.=20
=0A=
           4.4.7. Input Capture [7.5]=20
    =20
    P+: BEEP does not provide a mechanism for storing requests. Storing=20
    of input could be added by the server.=20
=0A=
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         4.5. Analysis of Speaker Identification and Verification=20
           Requirements=20
=0A=
           4.5.1. Requesting SI/SV [8.1]=20
    =20
    P+: An SI/SV resource would be defined as a new BEEP profile. To=20
    initiate an SI/SV session, the client would need to ask for this=20
    profile as part of the <start> message which opens a new channel.=20
=0A=
           4.5.2. Identifiers for SI/SV [8.2]=20
    =20
    T: This could be handled in the 'Content-Type' or by other means.=20
    Supported formats might be negotiated in a fashion similar to=20
    security negotiation.=20
=0A=
           4.5.3. State for multiple utterances [8.3]=20
    =20
    P+: BEEP has a concept of sequence. State would need to be added by=20
    the client. This should be easy if a channel is used for multiple=20
    utterances.=20
=0A=
           4.5.4. Input Capture [8.4]=20
    =20
    P+: BEEP does not provide a mechanism for storing requests. Storing=20
    of input could be added by the server.=20
=0A=
           4.5.5. SI/SV functional extensibility [8.5]=20
    =20
    T: Extensions could be added as additional messages.=20
=0A=
         4.6. Analysis of Duplexing and Parallel Operation Requirements=20
=0A=
           4.6.1. Duplexing and Parallel Operation Requirements [9]=20
    =20
    P+: Parallel operations may be obtained using multiple channels. A=20
    message on one channel could potentially interrupt activity=20
    happening on the second. BEEP is very flexible allowing the server=20
    to implement whatever behavior is desired.=20
=0A=
           4.6.2. Full Duplex operation [9.1.1]=20
    =20
    T: BEEP is a peer-to-peer protocol allowing full duplex=20
    communication on a single channel or parallel communication on=20
    multiple channels.=20
=0A=
           4.6.3. Multiple services in parallel [9.1.2]=20
    =20
    P+: Multiple services may be run on separate channels. Merging or=20
    T-ing of RTP must be implemented by the server.=20
=0A=
           4.6.4. Combination of services=20
    TBD=20
=0A=
         4.7. Analysis of additional considerations (non-normative)=20
    TBD=20
=0A=
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         4.8. Analysis of Security considerations=20
=0A=
           4.8.1. Security Considerations [11]=20
    =20
    P+: BEEP offers a mechanism for managing security and user=20
    authentication. =20
    SpeechSC requires managing multiple data streams and some form of=20
    unified authentication / security might be a goal. If so, BEEP=20
    security should be revisited with this in mind.=20
    =20
    =20
      5. =0D         Protocol =93SIP=94 Complience Evaluation (Rajiv =
Dharmadhikari)=20
=0A=
         5.1. Introduction=20
    SIP is a protocol for initiating, modifying, and terminating=20
    multimedia sessions. The protocol is considered an IETF standard=20
    and its specifications can be found in [2]. The following sections=20
    provides a general statement with regards to the applicability of=20
    SIP as the control protocol for SPEECHSC. =20
    =20
=0A=
           5.1.1. SIP General Applicability=20
    SIP is a pretty mature, well understood, and frequently used =20
    session establishment protocol. It has gone through multiple=20
    revisions in the IETF standard process. There are number of=20
    commercial and public domain implementations of SIP that are=20
    available. Because of its close resemblance to HTTP and being a=20
    text based protocol, there are large number of SIP application=20
    developers available. =20
    =20
=0A=
           5.1.2. SIP Use in VOIP environment=20
    SIP is already being used to establish and redirect RTP streams=20
    from various end points. The SPEECHSC requires a protocol for=20
    controlling ASR, TTS and SV resources. When these resources are=20
    deployed in a VOIP network that requires them to process media=20
    carried in RTP, the SIP protocol is used in lot of deployments.=20
    Rather than inventing a new control protocol and introducing=20
    operational aspects of the new protocol, SIP can be reused for=20
    controlling SPEECHSC resources. =20
    =20
=0A=
         5.2. Analysis of General Requirements=20
=0A=
           5.2.1. Reuse existing protocols [5.1]=20
    T: SIP is an existing, widely used, and mature protocol defined in=20
    [2].=20
=0A=
           5.2.2. Maintain Existing Protocol Integrity [5.2]=20
    T: Existing SIP methods and header fields will not be changed when=20
    SIP is used to control SPEECHSC resources. In case, if extensions=20
    are required, SIP allows carriage of custom payload in the body.=20
    This payload is understood only by UAs and it does not impact=20
    protocol integrity.=20
    =20
=0A=
=0A=
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           5.2.3. Avoid Duplicating Existing Protocols [5.3]=20
    T: Lot of the requirements for SPEECHSC operation can easily be=20
    satisfied by SIP, e.g. establishing RTP streams or redirecting=20
    them. Without SIP, new SPEECHSC protocol will have to duplicate lot=20
    of session management functionality.=20
    =20
=0A=
           5.2.4. Protocol efficiency [5.4]=20
    T: SIP is a very light weight protocol when run over TCP or UDP. It=20
    leverages efficiency available in TCP and UDP protocols that have=20
    been around for over 20 years.=20
=0A=
           5.2.5. Explicit invocation of services [5.5]=20
    T: SIP URI mechanism allows invocation of different services.=20
=0A=
           5.2.6. Server Location and Load Balancing [5.6]=20
    P+: SIP employs standard DNS name resolution for locating=20
    resources. SIP itself does not provide load balancing features.=20
    Application level load balancers can be used to load balance SIP=20
    requests.=20
    =20
=0A=
           5.2.7. Simultaneous services [5.7]=20
    T: SIP allows simultaneous invocation of different services. SIP=20
    allows forking or splitting the same media stream to different end=20
    points as defined in [2].=20
    =20
=0A=
           5.2.8. Multiple media sessions [5.8]=20
    T: SIP uses SDP to describe RTP stream characteristics. This allows=20
    the control of direction of RTP stream such as bi-directional or=20
    uni-directional. SIP allows a UA to establish sessions with=20
    multiple UAs for the same session.=20
=0A=
         5.3. Analysis of TTS requirements=20
=0A=
           5.3.1. Requesting Text Playback [6.1]=20
    P+: SIP does not provide primitives for text playback. New SIP URI=20
    can be defined to invoke TTS features. =20
=0A=
           5.3.2. Text Formats [6.2]=20
    T: SIP message body can carry TTS text or reference to the TTS=20
    text. =20
    =20
=0A=
           5.3.3. Plain text [6.2.1]=20
    T: =20
=0A=
           5.3.4. SSML [6.2.2]=20
    T:=20
=0A=
           5.3.5. Text in Control Channel [6.2.3]=20
    T :=20
=0A=
           5.3.6. Document Type Indication [6.2.4]=20
    T:=20
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           5.3.7. Control Channel [6.3]=20
    T: Separate SIP URI can be defined for control channel. =20
=0A=
           5.3.8. Playback Controls [6.4]=20
    P+: See 5.3.1=20
=0A=
           5.3.9. Session Parameters [6.5]=20
    P+: Session parameters can be passed in body part of SIP or can be=20
    modified by using mid call methods of SIP.=20
=0A=
           5.3.10. =0D                  Speech Markers [6.6]=20
    =20
    T: Speech markers can be delivered using a separate control=20
    channel. See 5.3.7.=20
=0A=
         5.4. Analysis of ASR requirements=20
=0A=
           5.4.1. Requesting Automatic Speech Recognition [7.1]=20
    T: New SIP URI can be defined to address ASR services and=20
    requesting ASR resources.=20
=0A=
           5.4.2. XML [7.2]=20
    T: SIP message body can carry XML.=20
=0A=
           5.4.3. Grammar Specification [7.3.1]=20
    P+: With proper definition, SIP message body can carry XML grammars=20
    or references to the XML grammars.=20
 =20
=0A=
           5.4.4. Explicit Indication of Grammar Format [7.3.2]=20
    P+: See 5.4.3.=20
=0A=
           5.4.5. Grammar Sharing [7.3.3]=20
    TBD=20
=0A=
           5.4.6. Session Parameters [7.4]=20
    T: Session parameters can be passed in body part of SIP or can be=20
    modified by using mid call methods of SIP.=20
=0A=
           5.4.7. Input Capture [7.5]=20
    T: SIP SUBSCRIBE/NOTIFY mechanism can be used initiate input=20
    capture and receive the result. =20
    =20
=0A=
         5.5. Analysis of Speaker Identification and Verification=20
           Requirements=20
=0A=
           5.5.1. Requesting SI/SV [8.1]=20
    T: New SIP URI can be defined for SI/SV.=20
=0A=
           5.5.2. Identifiers for SI/SV [8.2]=20
    T: See 5.5.1.=20
=0A=
           5.5.3. State for multiple utterances [8.3]=20
    TBD=20
=0A=
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           5.5.4. Input Capture [8.4]=20
    T: SIP SUBSCRIBE/NOTIFY mechanism can be used initiate input=20
    capture and receive the result.=20
=0A=
           5.5.5. SI/SV functional extensibility [8.5]=20
    T: SIP can easily be extended by adding new methods and header=20
    fields without impacting the operation of existing methods. The new=20
    methods and headers will only be understood by the SPEECHSC=20
    entities. If above is not acceptable, SIP message body of an=20
    existing primitive can be used to define new semantics that is=20
    understood only by SPEECHSC entities.=20
=0A=
         5.6. Analysis of Duplexing and Parallel Operation Requirements=20
=0A=
           5.6.1. Duplexing and Parallel Operation Requirements [9]=20
    T: SPEECHSC resource is a SIP UA that can handle session requests=20
    from different UAs.=20
=0A=
           5.6.2. Full Duplex operation [9.1.1]=20
    T: Each SIP UA consists of a UAC and a UAS. This allows for full=20
    duplex operation.=20
=0A=
           5.6.3. Multiple services in parallel [9.1.2]=20
    T: SIP allows simultaneous invocation of different services. SIP=20
    allows forking or splitting the same media stream to different end=20
    points as defined in [2].=20
    =20
=0A=
           5.6.4. Combination of services=20
    T: See 5.6.3. SIP UA can invoke different services and combine the=20
    results.=20
=0A=
         5.7. Analysis of additional considerations (non-normative)=20
    TBD=20
=0A=
         5.8. Analysis of Security considerations=20
=0A=
           5.8.1. Security Considerations [11]=20
    T: SIP protocol employs different authentication schemes that are=20
    widely used in IP based protocols.=20
=0A=
         5.9. Other Criteria=20
    The following criteria were also defined by the evaluator of SIP.=20
=0A=
           5.9.1. Ability to establish session between SPEECHSC client=20
              and SPEECHSC resource=20
    T: SIP User Agent can establish a session with another SIP User=20
    Agent.=20
=0A=
           5.9.2. Ability to terminate session by either SPEECHSC=20
              client or SPEECHSC resource=20
    T: SIP User Agent can terminate a session with another SIP User=20
    Agent.=20
=0A=
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           5.9.3. Support reliable sequencing and delivery between=20
              SPEECHSC client and SPEECHSC resource=20
    P: SIP can be run over TCP or UDP. When run over TCP, this=20
    requirement is easily satisfied. When run over UDP, SIP User Agent=20
    is required to implement logic to ensure reliable sequencing and=20
    delivery. =20
=0A=
           5.9.4. Ability for SPEECHSC client to coordinate SPEECHSC=20
              resources on different machines for a single session=20
    T: SPEECHSC client can use SIP to establish SIP sessions with=20
    different machines.=20
=0A=
           5.9.5. Ability for SPEECHSC resource to handle multiple=20
              SPEECHSC clients=20
    T: SPEECHSC resource is a SIP UA that can handle session requests=20
    from different UAs.=20
=0A=
           5.9.6. The SPEECHSC resource should be able to generate=20
              asynchronous events or unsolicited messages=20
    T: SIP allows asynchronous events or unsolicited messages to be=20
    generated using SUBSCRIBE/NOTIFY mechanism.=20
=0A=
           5.9.7. The SPEECHSC client and resource should have ability=20
              for authenticating each other=20
    T: SIP protocol employs different authentication schemes that are=20
    widely used in IP based protocols.=20
=0A=
           5.9.8. Ability to determine success or failure from both=20
              SPEECHSC client and SPEECHSC resource side=20
    T: The protocol has following response codes: 200 for success, 3xx,=20
    4xx, and 5xx for failure.=20
=0A=
           5.9.9. Support for versioning between SPEECHSC client and=20
              SPEECHSC resource=20
    P+: This will require an additional header or element in the   =20
    body of SIP message for versioning. The current version field is=20
    intended for SIP protocol version. =20
    =20
    =20
      6. =0D         Protocol =93RTSP=94 Complience Evaluation (Brian =
Wyld)=20
=0A=
         6.1. General Introduction=20
    RTSP is an existing protocol, orientated towards audio playback and=20
    recording. As such, it has support for RTP session control, with=20
    SDP used for session description, and a message set allowing=20
    operation as a player/recorder with audio =93VCR=94 controls.=20
    =20
    To use RTSP for speechsc, two options present themselves:=20
     - new =93operations=94 (like PLAY) would be defined with their own=20
        specific state machines after a session is created. Note that=20
        this could be seen as the merge of the MRCP proposed protocol=20
        operations into the RTSP protocol directly.=20
     - The existing operations could be =91extended=92 to describe =
speechsc=20
        operations eg PLAY would both be applicable for play of a audio=20
        file and for synthesis of a TTS text. =20
     =20
    In the second case new headers would be defined to allow definition=20
    of the text or of its location. The current PLAY state machine is=20
 =20
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    exactly as required for TTS operation. Although by analogy RECORD=20
    could initiate an ASR session, with headers giving the grammer=20
    source or references, it=92s state machine does not seem as=20
    compatible, still less for SV/SI. Defining new messages/state=20
    machines would be better for these operations.=20
    =20
    The following analysis assumes that TTS operation is merged with=20
    the current PLAY semantics, while ASR/SV/SI create new messages and=20
    state operations.=20
    =20
    The following sub-sections compare each individual requirement=20
    against the protocol.=20
=0A=
         6.2. Analysis of General Requirements=20
=0A=
           6.2.1. Reuse existing protocols [5.1]=20
    T: RTSP/RTP/SDP would be reused.=20
=0A=
           6.2.2. Maintain Existing Protocol Integrity [5.2]=20
    T: The extensions to RTSP to allow speechsc use would be in the=20
    spirit of the protocol, and would not break existing servers or=20
    clients.=20
=0A=
           6.2.3. Avoid Duplicating Existing Protocols [5.3]=20
    T: Using RTSP would not recreate it.=20
=0A=
           6.2.4. Protocol efficiency [5.4]=20
    T: RTSP is a text based protocol, but is relatively succinct as=20
    messages are specific to their operation.=20
=0A=
           6.2.5. Explicit invocation of services [5.5]=20
    T: RTSP service invocation is sufficient.=20
=0A=
           6.2.6. Server Location and Load Balancing [5.6]=20
    F: RTSP does not address this topic; however it can be used with=20
    other IETF protocols such as SLP or UDDI to do so.=20
=0A=
           6.2.7. Simultaneous services [5.7]=20
    T: RTSP allows simultaneous invocation of services on the same or=20
    different control channel.=20
=0A=
           6.2.8. Multiple media sessions [5.8]=20
    T: RTSP allows multiple media sessions.=20
=0A=
         6.3. Analysis of TTS requirements=20
=0A=
           6.3.1. Requesting Text Playback [6.1]=20
    P+: by extension of the RTSP PLAY message semantics=20
=0A=
           6.3.2. Text Formats [6.2]=20
    P+: Text can be defined as all text types. =20
=0A=
           6.3.3. Plain text [6.2.1]=20
    T: Plain text may be carried directly in the message payload.=20
=0A=
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           6.3.4. SSML [6.2.2]=20
    T: Text may be in any format.=20
=0A=
           6.3.5. Text in Control Channel [6.2.3]=20
    T: Text may be attached to the control messages.=20
=0A=
           6.3.6. Document Type Indication [6.2.4]=20
    T : Via the Content-Type header=20
=0A=
           6.3.7. Control Channel [6.3]=20
    T: RTSP sessions may use a private or shared TCP connection.=20
=0A=
           6.3.8. Playback Controls [6.4]=20
    T: RTSP defines playback control messages and a state machine.=20
=0A=
           6.3.9. Session Parameters [6.5]=20
    T: RTSP defines operations for session parameter control.=20
=0A=
           6.3.10. =0D                  Speech Markers [6.6]=20
    P+: Markers may be inserted in the text, but to provide the=20
    required asynchronous events when a marker is synthesized will=20
    require use specific ANNOUNCE type messages for server->client=20
    notification.=20
=0A=
         6.4. Analysis of ASR requirements=20
=0A=
           6.4.1. Requesting Automatic Speech Recognition [7.1]=20
    P+: by addition of a message and state machine.=20
=0A=
           6.4.2. XML [7.2]=20
    P+: Text can be defined as all text types.=20
=0A=
           6.4.3. Grammar Specification [7.3.1]=20
    P+: Text can be defined as all text types.=20
=0A=
           6.4.4. Explicit Indication of Grammar Format [7.3.2]=20
    T : Via the Content-Type headers=20
=0A=
           6.4.5. Grammar Sharing [7.3.3]=20
    F: TBD=20
=0A=
           6.4.6. Session Parameters [7.4]=20
    T: RTSP defines operations for session parameter control.=20
=0A=
           6.4.7. Input Capture [7.5]=20
    P+: addition of a header to the initiation message.=20
=0A=
         6.5. Analysis of Speaker Identification and Verification=20
           Requirements=20
=0A=
           6.5.1. Requesting SI/SV [8.1]=20
    P+: by addition of a message and state machine.=20
=0A=
           6.5.2. Identifiers for SI/SV [8.2]=20
    P+: by addition of specific headers.=20
 =20
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           6.5.3. State for multiple utterances [8.3]=20
    TBD=20
=0A=
           6.5.4. Input Capture [8.4]=20
    P+: addition of a header to the initiation message.=20
=0A=
           6.5.5. SI/SV functional extensibility [8.5]=20
    TBD=20
=0A=
         6.6. Analysis of Duplexing and Parallel Operation Requirements=20
=0A=
           6.6.1. Duplexing and Parallel Operation Requirements [9]=20
    T: RTSP allows session setup that should fulfill these=20
    requirements.=20
=0A=
           6.6.2. Full Duplex operation [9.1.1]=20
    T: RTSP can create a full duplex session.=20
=0A=
           6.6.3. Multiple services in parallel [9.1.2]=20
    T: RTSP can request multiple operations of the same type on the=20
    same session.=20
=0A=
           6.6.4. Combination of services=20
    T: RTSP can request multiple operations of different types on the=20
    same session.=20
=0A=
         6.7. Analysis of additional considerations (non-normative)=20
    TBD=20
=0A=
         6.8. Analysis of Security considerations=20
=0A=
           6.8.1. Security Considerations [11]=20
    F: RTSP provides no specific security functionality at all, but=20
    depends on other IETF security protocols (as it uses TCP) to pre-
    validate and protect the sessions.=20
    =20
      7. =0D         Protocol =93MRCP=94 Complience Evaluation (Sarvi =
Shanmugham)=20
=0A=
         7.1. General =20
=0A=
           7.1.1. MRCP Framework and General Applicability=20
    =20
    The overall MRCP framework, the components involved and their=20
    distribution and relationship to each other meet the framework=20
    specified by SPEECHSC. The primary advantage of MRCP is that it is=20
    a text based protocol designed to meet most of the requirements of=20
    SPEECHSC pertaining to speech recognition and Text to speech.=20
    Though Speaker Recognition (SR) and Speaker Verification (SV) are=20
    not supported in its current form, MRCP was explicitly designed to=20
    be extendable for such needs. The core MRCP definition only deals=20
    with the control of the ASR or TTS resource and the commands and=20
    responses needed to achieve it.=20
 =20
    There are multiple interoperable implementations of MRCP and hence=20
    is a proven technology. It leverages existing W3C XML standards for=20
    exchange of data between the client and the server resource. For=20
 =20
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    Example, its uses the W3C XML grammar format (GRXML) along with W3C=20
    semantic attachments and Natural Language Semantic Markup Language=20
    to exchange data with speech recognition resource. The W3C Speech=20
    Markup Language is used when dealing with Text to speech engines.=20
     =20
    It was designed to work as a tunneled protocol, over RTSP or SIP.=20
    Hence it depends on the carrier protocol to establish a control and=20
    a media path between the client and the ASR or TTS server resource.=20
    Hence it gets most of the security and media pipe management=20
    operations for free. Once these are established, MRCP commands and=20
    responses are tunneled over, controlling the ASR or TTS resource on=20
    the server. =20
=0A=
           7.1.2. MRCP can be evolved=20
    =20
    Though MRCP directly meets many of the needs of SPEECHSC. The=20
    notion that it is a tunneled protocol disallows its independent=20
    operation. Further more the tunneled aspect is also a less=20
    efficient protocol design. =20
    =20
    But these can be addressed and the core MRCP messages can be=20
    evolved to either become standalone protocol by itself or=20
    extensions to an existing protocol such as SIP or RTSP.  To make=20
    this a standalone protocol and allow MRCP to operate by itself, new=20
    session and media management messages need to be defined to allow=20
    it to operate independently. To evolve MRCP as extensions to SIP or=20
    RTSP would also be relatively simple since it is also a text based=20
    protocol with message format and headers very similar to them.  In=20
    this protocol evaluation, the compliance evaluates MRCP from the=20
    perspective of evolution in one of these forms.=20
    =20
    The following sub-sections compare each individual requirement=20
    against the protocol.=20
=0A=
         7.2. Analysis of General Requirements=20
=0A=
           7.2.1. Reuse existing protocols [5.1]=20
    T: If RTSP or SIP is extended with MRCP, it will be reusing an=20
    existing protocol. If MRCP was extended to become a standalone=20
    protocol, it wouldn=92t.=20
=0A=
           7.2.2. Maintain Existing Protocol Integrity [5.2]=20
    T: If RTSP or SIP is extended with MRCP, it can be done in a way=20
    that maintains the integrity of the protocol.=20
=0A=
           7.2.3. Avoid Duplicating Existing Protocols [5.3]=20
    T: If RTSP or SIP is extended with MRCP, it would be meet this=20
    requirement. If MRCP was extended to become a standalone protocol,=20
    it wouldn=92t.=20
=0A=
           7.2.4. Protocol efficiency [5.4]=20
    P: MRCP as it exists is sub-optimal due to its tunneling. On the=20
    other hand, if it were extended as standalone protocol or with RTSP=20
    or SIP, it would meet this requirement.=20
=0A=
           7.2.5. Explicit invocation of services [5.5]=20
    T:=20
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           7.2.6. Server Location and Load Balancing [5.6]=20
    T:If MRCP were extended as a standalone protocol it could designed=20
    to meet this requirement. If MRCP was used to extend SIP or RTSP,=20
    it would automatically inherit their capability this area and hence=20
    would meet this requirement.=20
=0A=
           7.2.7. Simultaneous services [5.7]=20
    T: MRCP as it exists does meet this requirement already. It could=20
    continue to do so even if it is extended in wither of the ways=20
    suggested here.=20
=0A=
           7.2.8. Multiple media sessions [5.8]=20
    P+: MRCP shares a single 2 way media pipe between ASR and TTS=20
    today. Considering it supports only 2 resources and they treat=20
    media in different directions it currently doesn=92t meet this=20
    requirement completely. Again, SI or SV was added to MRCP as it=20
    exists, a standalone protocol or as an extension to RTSP/SIP, it=20
    could be designed to meet this requirement.=20
=0A=
         7.3. Analysis of TTS requirements=20
=0A=
           7.3.1. Requesting Text Playback [6.1]=20
    T: MRCP has the SPEAK method for the client to request the TTS=20
    resource to playback text as an audio stream.=20
=0A=
           7.3.2. Text Formats [6.2]=20
    T: When the client requests the TTS resource to playback a text=20
    stream it can provide the content in the following formats and=20
    through the following mechanism.=20
    =20
       1. Plain text=20
       2. W3C XML based Speech Markup Language (SSML)=20
       3. This content to be spoken can be provided by value directly=20
          through the control path. =20
       4. It also supports passing the content by reference. This is=20
          achieved having an audio tag inside the SSML markup text.=20
          This URL is then fetched and played on the RTP stream in=20
          sequence with the rest of the text according to the SSML=20
          specification.=20
    When the client sends plain text, SSML or another format of speech=20
    text the content is coded as a mime-type. Hence the server knows=20
    what format the speech content is coded in, and does not have to=20
    figure it out from the content.=20
=0A=
           7.3.3. Plain text [6.2.1]=20
    T: see above=20
=0A=
           7.3.4. SSML [6.2.2]=20
    T: see above=20
=0A=
           7.3.5. Text in Control Channel [6.2.3]=20
    T : see above=20
=0A=
           7.3.6. Document Type Indication [6.2.4]=20
    T: see above=20
=0A=
=0A=
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           7.3.7. Control Channel [6.3]=20
    T: In MRCP, this Reset-Audio-Channel header defined for the ASR=20
    resource allows the recognizer to re-initialize the audio=20
    characteristics that it has learnt till then. This allows a=20
    recognizer resource to be used for multiple recognition sessions.=20
    It can be used for short single utterance recognitions as well.=20
    This is by applying the Reset-Audio-Channel header to every=20
    recognition. I suspect the performance may not be as good, due to=20
    the lack of line characteristics, but this is a recognizer issue.=20
=0A=
           7.3.8. Playback Controls [6.4]=20
    T: MRCP supports the CONTROL method with the Jump-Target header can=20
    used to achieve, jumping in time or to an exact or relative=20
    location. It supports jumping in paragraphs, sentences, words and=20
    to specific markers that may be embedded in the speech content. The=20
    CONTROL method can be used with the Voice and Prosody parameters,=20
    derived from SSML, and can address the speed of speech or=20
    increasing/decreasing the volume. It also supports the PAUSE/RESUME=20
    methods to pause or resume a current SPEAK request.=20
=0A=
           7.3.9. Session Parameters [6.5]=20
    T: As mentioned the previous section, MRCP supports voice and=20
    prosody parameters which are directly derived from the W3C SSML=20
    specification. These headers can be sent using the SET-PARAMS=20
    method and applied as a default for the entire session. They can=20
    also be applied in SPEAK requests to apply per usage or in the=20
    CONTROL message to change the parameters of an active SPEAK=20
    request.=20
=0A=
           7.3.10. =0D                  Speech Markers [6.6]=20
    T: Specifying speech markers in the content is supported through=20
    SSML. The CONTROL message can then be used to jump to specific=20
    marker points in the text. Also, when the TTS resource reaches=20
    specific markers in the text, the server would generate the SPEECH-
    MARKER method to the client.=20
=0A=
         7.4. Analysis of ASR requirements=20
=0A=
           7.4.1. Requesting Automatic Speech Recognition [7.1]=20
    T: The client uses the RECOGNIZE method in MRCP to request the=20
    recognition resource to process the audio stream in the pipe. The=20
    RECOGNIZE method also specifies parameters and grammars the=20
    recognizer should match against.=20
=0A=
           7.4.2. XML [7.2]=20
    T: Similar to the TTS resource in MRCP, ASR also uses XML data to=20
    exchange information between the client and the recognition=20
    resource. It supports the W3C GRXML to pass grammars from the=20
    client to the server. When the server is done recognizing, it uses=20
    the W3C Natural Language Semantic Markup Language (NLSML) to pass=20
    the results back to the client. It supports other grammar formats=20
    as well, as long as the server allows it. This is possible since,=20
    it uses mime-types to package this data and hence the format type=20
    is specified.=20
=0A=
           7.4.3. Grammar Specification [7.3.1]=20
    P+: MRCP supports specifying the grammar both by value and by=20
    reference. The RECOGNIZE method can carry with it grammar content=20
 =20
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    and/or a URI referring to the grammar content. Since MRCP supports=20
    referring a grammar, the referred grammar could be located on the=20
    server itself. With respect to sharing of grammars, the grammars=20
    defined/compiled through the DEFINE-GRAMMAR primitive are not=20
    sharable across sessions on the same server. This needs to be=20
    addressed to meet this set of requirements in full.=20
=0A=
           7.4.4. Explicit Indication of Grammar Format [7.3.2]=20
    P+: see above=20
=0A=
           7.4.5. Grammar Sharing [7.3.3]=20
    TBD=20
=0A=
           7.4.6. Session Parameters [7.4]=20
    T: This requirement as defined is already fully met since MRCP is=20
    the referred standard for compliance.=20
=0A=
           7.4.7. Input Capture [7.5]=20
    T: This is achieved by setting the Waveform-url header in the=20
    RECOGNIZE method. This tells the server to record the audio of the=20
    recognition and will return a URI to the client in the completion=20
    event, which can be used to retrieve or play back the audio.=20
=0A=
         7.5. Analysis of Speaker Identification and Verification=20
           Requirements=20
=0A=
           7.5.1. Requesting SI/SV [8.1]=20
    F: not supported=20
=0A=
           7.5.2. Identifiers for SI/SV [8.2]=20
    F: not supported=20
=0A=
           7.5.3. State for multiple utterances [8.3]=20
    F: not supported=20
=0A=
           7.5.4. Input Capture [8.4]=20
    F: not supported=20
=0A=
           7.5.5. SI/SV functional extensibility [8.5]=20
    F: not supported=20
=0A=
         7.6. Analysis of Duplexing and Parallel Operation Requirements=20
=0A=
           7.6.1. Duplexing and Parallel Operation Requirements [9]=20
=0A=
           7.6.2. Full Duplex operation [9.1.1]=20
    T: MRCP supports having the ASR and TTS engines on the same server.=20
    When in this mode they use a single full-duplex audio stream to=20
    both recognize audio and generate a TTS voice stream.=20
    =20
=0A=
           7.6.3. Multiple services in parallel [9.1.2]=20
    F: MRCP doesn=92t support SI or SV.=20
=0A=
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           7.6.4. Combination of services=20
    F: MRCP assumes that the recognizer to be a combined unit and does=20
    not support separating of sub-components like semantic analyzers.=20
=0A=
         7.7. Analysis of additional considerations (non-normative)=20
    T: Since MRCP works with SIP or RTSP, it uses SDP for session=20
    description and can work with other transport schemes like ATM etc.=20
    It uses standard codecs and can work with specialized codecs like=20
    DSR.=20
=0A=
         7.8. Analysis of Security considerations=20
=0A=
           7.8.1. Security Considerations [11]=20
    T: Since MRCP works with RTSP, it inherits the security=20
    considerations of SIP or RTSP.=20
    =20
    =20
      8. =0D         Protocol =93Web Services=94 Complience Evaluation =
(Stephane H.=20
         Maes)=20
=0A=
         8.1. General Notes:=20
    Speech engines (speech recognition, speaker, recognition, speech =20
    synthesis, recorders and playback, NL parsers, and any other speech=20
    processing engines (e.g. speech detection, barge-in detection etc) =20
    etc...) as well as audio sub-systems (audio input and output =20
    sub-systems) can be considered as web services that can be =20
    described and asynchronously programmed via WSDL (on top of SOAP), =20
    combined in a flow described via WSFL, discovered via UDDI and =20
    asynchronously controlled via SOAP that also enables =20
    asynchronous exchanges between the engines.=20
     =20
    This solution presents the advantage to provide flexibility, =20
    scalability and extensibility while reusing an existing framework =20
    that fits the evolution of the web: web services and XML protocols=20
    [WS1]=20
    =20
    According to the web services framework, speech engines (audio =20
    sub-systems, engines, speech processors) can be defined as web=20
    services=20
    that are characterized by an interface that consists of some of the  =

    following ports:=20
        - "control in" port(s): It sets the engine context, i.e. all=20
    the=20
        settings required for a speech engine to run. It may include =20
        addresses where to get or send the streamed audio or results.=20
        - "control out" port(s): It produces the non-audio engine=20
    output=20
        (i.e. results and events). It may also involve some session =20
        control exchanges.=20
        - "audio in" port(s): It receives streamed input data. =20
        - "audio out" port(s): It produces streamed output data. =20
    =20
    Audio sub-systems can also be treated as web services that can =20
    produce streamed data or play incoming streamed data as specified=20
    by=20
    the control parameters.=20
    =20
    The "control in" or "control out" messages can be out-of-band or =20
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    sent or received interleaved with "audio in or out" data. This can =20
    be determined in the context (setup) of the web services. =20
    =20
    Speech engines and audio sub-systems are pre-programmed as web =20
    services and composed into more advanced services. Once programmed =20
    by the application / controller, audio-sub-systems and engines=20
    await=20
    an incoming event (established audio session, etc...) to execute=20
    the=20
    speech processing that they have been programmed to do and send the  =

    results as programmed. =20
    =20
    Speech engines as web services are typically programmed to handle =20
    completely a particular speech processing task, including handling =20
    of possible errors. For example, as speech engine is programmed to =20
    perform recognition of the next incoming utterance with a=20
    particular=20
    grammar, to send result to a NL parser and to contact a particular =20
    error recovery process if particular errors occur.=20
    =20
    The following sub-sections compare each individual requirement=20
    against the protocol.=20
=0A=
         8.2. Analysis of General Requirements=20
=0A=
           8.2.1. Reuse existing protocols [5.1]=20
    T: Web services are is a class of protocols (framework) widely=20
    studied and developed across numerous standard bodies like W3C,=20
    OASIS, WS-I, Liberty, Parlay and adapted to numerous deployment=20
    environments  issues at IETF, OMA, 3GPP, 3GPP2, JCP, etc=85 As an=20
    entry point, we recommend consulting the work at W3C [WS1].=20
    =20
=0A=
           8.2.2. Maintain Existing Protocol Integrity [5.2]=20
    T: Web services is an XML-based framework that is by definition=20
    extensible to support appropriate syntax and semantics. =20
    =20
    Web services are bound on underlying transport protocols. Numerous=20
    such binding have been specified. Others are in development. By=20
    handling at SPEECHSC at the level of the =20
    Web services framework, the integrity is maintained for:=20
    - underlying transport protocols (to which the web service are=20
    bound (e.g. SOAP)=20
    - web service framework=20
    =20
    This does not prevent introducing bindings to new protocols if=20
    needed. For example, binding to SIP or BEEP could be advantageous=20
    for mobile deployments.=20
    =20
=0A=
           8.2.3. Avoid Duplicating Existing Protocols [5.3]=20
    T: By definition, the web service framework can be specified to=20
    remote control any web service. Specified syntax can be limited to=20
    avoid duplicating remote control functionalities offered by other=20
    protocols. =20
    =20
    At the same time, the extensibility inherent to the framework=20
    guarantees that it is possible to specify (standard) or define=20
=0A=
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    (application specific) remote control for other entities beyond the=20
    current scope of SPEECHSC. =20
    =20
    In that context and in view of unifying the remote control=20
    framework exposed to an application developer or a system=20
    integrator, it may be of interest to provide remote control syntax=20
    for special entities like prompt player etc=85=20
    =20
=0A=
           8.2.4. Protocol efficiency [5.4]=20
    P+ to P: Web services are by definition more verbose protocols.=20
    Hence, at this stage this does not qualify work a T mark. =20
    =20
    However work is in progress (e.g. OMA, JCP) to optimize the=20
    exchanges to handle:=20
    - Client with limited resources=20
    - Constrained bandwidth=20
    These rely on protocol compression and optimization, caching and=20
    gateways. =20
    =20
    As such the protocols qualify as P+.=20
    =20
    In addition, based on the qualification of efficiency provided in=20
    [WS8], the web service framework proposed for SPEECHSC and=20
    described in [WS1] relies indeed on known efficient techniques:=20
    - Asynchronous pre-programming of the engines as web services to=20
    reduce exchanges and avoid racing conditions=20
    - Possibility to piggy back on response message if transported on=20
    optimized protocols like SIP or BEEP. =20
    - state caching in the engines that are considered as stand-alone,=20
    pre-packaged and pre-programmed engines.=20
    - etc=85=20
    =20
=0A=
           8.2.5. Explicit invocation of services [5.5]=20
    T: Web service is typically used in a client-server environment.=20
    Solutions exist for peer to peer (service to service) etc=85 =20
    =20
    Web services have been deigned to support clients and servers at=20
    least one of which is operating directly on behalf of the user=20
    requesting the service.=20
    =20
    In addition, work on-going at OMA and JCP addresses some of these=20
    issues in mobile environment with the introduction of possible web=20
    service gateways. =20
    =20
=0A=
           8.2.6. Server Location and Load Balancing [5.6]=20
    T: Web services are widely developed for e-business applications.=20
    Numerous tools and mechanisms have been provided for service=20
    discovery ad advertisement. In addition, numerous offerings provide=20
    routing and load balancing capabilities as part of the web=20
    application server used to deploy the web service. =20
    =20
    Note that web services do not specify server location or load=20
    balancing; but they are deployed on systems that provide such=20
    functionalities. As web services are expected to be widely used in=20
    the future and central to most e-business offerings, it is to=20
=0A=
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    expect that such tools will become even more pervasive and=20
    efficient.=20
    =20
=0A=
           8.2.7. Simultaneous services [5.7]=20
    Web services allow control (interface) and composition of web=20
    services at will (e.g. WSFL).=20
=0A=
           8.2.8. Multiple media sessions [5.8]=20
    T: The framework proposed does not pre-supposes how many ports or=20
    streams are associated to the engine. Different inbound and=20
    outbound can be used at will=20
=0A=
         8.3. Analysis of TTS requirements=20
=0A=
           8.3.1. Requesting Text Playback [6.1]=20
    T: (supported =96 syntax to be defined; which is consistent with the =

    web service framework)=20
    =20
    As described, TTS engines can be pre-programmed as web services to=20
    perform TTS on incoming text. This is simply a matter of agreeing=20
    on the control syntax to do so. The text to play back can be part=20
    of the control instructions transmitted in SOAP to the TTS engine.=20
=0A=
           8.3.2. Text Formats [6.2]=20
    T: Exchanged format for text can be any MIME type; including plain=20
    text.=20
=0A=
           8.3.3. Plain text [6.2.1]=20
    T: Exchanged format for text can be any MIME type; including plain=20
    text.=20
=0A=
           8.3.4. SSML [6.2.2]=20
    T: Exchanged format for text can be any MIME type; including XML=20
    and hence SSML=20
=0A=
           8.3.5. Text in Control Channel [6.2.3]=20
    TBD=20
=0A=
           8.3.6. Document Type Indication [6.2.4]=20
    T: SOAP and the web service framework built on SOAP rely on XML and=20
    MIME type to identify media types. This is at the core of data=20
    exchange in SOAP.=20
    =20
=0A=
           8.3.7. Control Channel [6.3]=20
    T: As proposed above, SOAP [WS2] and WSDL [WS3] support the remote=20
    control of the web services (engines or media processing entity).=20
=0A=
           8.3.8. Playback Controls [6.4]=20
    T: (supported =96 syntax to be defined; which is consistent with the =

    web service framework)=20
    =20
    This is simply a matter of agreeing on the control syntax to do so=20
    as part of the control instructions transmitted in SOAP to the TTS=20
    engine.=20
=0A=
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           8.3.9. Session Parameters [6.5]=20
    T: Session parameters are presumably content delivered as part of=20
    the control instructions transmitted in SOAP to the TTS engine.=20
    =20
=0A=
           8.3.10. =0D                  Speech Markers [6.6]=20
    T: Speech markers are presumably content delivered as part of the=20
    control instructions transmitted in SOAP to the TTS engine.=20
    [NDLR : how are the marker events returned to the client end?]=20
=0A=
         8.4. Analysis of ASR requirements=20
=0A=
           8.4.1. Requesting Automatic Speech Recognition [7.1]=20
    T: (supported =96 syntax to be defined; which is consistent with the =

    web service framework)=20
    =20
    As described, ASR engines can be pre-programmed as web services to=20
    perform speech recognition on incoming audio. This is simply a=20
    matter of agreeing on the control syntax to do so. The instructions=20
    and parameters (including data files like grammars etc=85) can be=20
    part of the control instructions transmitted in SOAP to the ASR=20
    engine. =20
    =20
    Results can be part of the web service messaging as supported by=20
    the web service framework.=20
=0A=
           8.4.2. XML [7.2]=20
    T: Exchanged format for message can be any MIME type; including XML=20
    and hence XML for controlling the ASR.=20
=0A=
           8.4.3. Grammar Specification [7.3.1]=20
    T: Grammar specification can be part of the messages to control the=20
    ASR. This includes any MIME type; including XML for passing=20
    grammars by values, other MIME format including binary and URI for=20
    passing grammars by reference.=20
=0A=
           8.4.4. Explicit Indication of Grammar Format [7.3.2]=20
    T: SOAP and the web service framework built on SOAP rely on XML and=20
    MIME type to identify media types. This is at the core of data=20
    exchange in SOAP.=20
=0A=
           8.4.5. Grammar Sharing [7.3.3]=20
    T: The framework described supports pre-programming of the engines=20
    per utterance, per session or in an unlimited manner. This way=20
    grammar sharing can easily be achieved and controlled by an=20
    external controller, application etc=85=20
=0A=
           8.4.6. Session Parameters [7.4]=20
    T: Session parameters are presumably content delivered as part of=20
    the control instructions transmitted in SOAP to the ASR engine.=20
=0A=
           8.4.7. Input Capture [7.5]=20
    T: (supported =96 syntax to be defined; which is consistent with the =

    web service framework)=20
    =20
    As described, ASR engines can be pre-programmed as web services to=20
    perform speech recognition on incoming audio. This is simply a=20
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    matter of agreeing on the control syntax to do so. The instructions=20
    and parameters (including data files like grammars etc=85) can be=20
    part of the control instructions transmitted in SOAP to the ASR=20
    engine. This cab include the syntax and instructions to capture the=20
    audio.=20
=0A=
         8.5. Analysis of Speaker Identification and Verification=20
           Requirements=20
=0A=
           8.5.1. Requesting SI/SV [8.1]=20
    T: (supported =96 syntax to be defined; which is consistent with the =

    web service framework)=20
    =20
    As described, SI or SV engines can be pre-programmed as web=20
    services to perform speaker recognition on incoming audio. This is=20
    simply a matter of agreeing on the control syntax to do so. The=20
    instructions and parameters (including data files like voice=20
    prints, etc=85) can be part of the control instructions transmitted=20
    in SOAP to the SI or SV engine. =20
    =20
    Results can be part of the web service messaging as supported by=20
    the web service framework.=20
=0A=
           8.5.2. Identifiers for SI/SV [8.2]=20
    T: This can be part of the control message.=20
=0A=
           8.5.3. State for multiple utterances [8.3]=20
    T: This can be achieved by appropriately programming the SI or SV=20
    engine across multiple utterances. This is simply a matter of=20
    agreeing on the control syntax to do so. The framework described in=20
    section 1 support spanning multiple utterances.=20
=0A=
           8.5.4. Input Capture [8.4]=20
    T: (supported =96 syntax to be defined; which is consistent with the =

    web service framework)=20
    =20
    As described, SI or SV engines can be pre-programmed as web=20
    services to perform speaker recognition on incoming audio. This is=20
    simply a matter of agreeing on the control syntax to do so. The=20
    instructions and parameters (including data files like grammars=20
    etc=85) can be part of the control instructions transmitted in SOAP=20
    to the ASR engine. This can include the syntax and instructions to=20
    capture the audio.=20
=0A=
           8.5.5. SI/SV functional extensibility [8.5]=20
    T: By definition a web service framework and XML are extensible to=20
    new functionality and describe how extensibility is achieved.=20
=0A=
         8.6. Analysis of Duplexing and Parallel Operation Requirements=20
=0A=
           8.6.1. Duplexing and Parallel Operation Requirements [9]=20
    T: As explained, web services allow control (interface) and=20
    composition of web services at will (e.g. WSFL).  Also, it does not=20
    pre-supposes how many ports or streams are associated to the=20
    engine. Different inbound and outbound can be used at will; in full=20
    duplex or even between engines as supported by WSFL [WS4] and WSXl=20
    [WS7].=20
=0A=
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           8.6.2. Full Duplex operation [9.1.1]=20
    T:=20
=0A=
           8.6.3. Multiple services in parallel [9.1.2]=20
    T:=20
=0A=
           8.6.4. Combination of services=20
    T: As explained, web services allow control (interface) and=20
    composition of web services at will (e.g. WSFL) into complex=20
    parallel, serial or coordinated combinations as supported by WSFL=20
    [WS4] and WSXl [WS7].=20
=0A=
         8.7. Analysis of additional considerations (non-normative)=20
    The framework proposed supports:=20
    - Use of SDP to describe sessions and streams for the streamed=20
    channels =20
    - Time stamps could be transmitted as part of the control messages=20
    at the web service level or in band (e.g. with dynamic payload=20
    switch or within the payload).=20
    - The framework is compatible with any encoding scheme. This is=20
    illustrated by the work on SRF (Speech Recognition Framework)=20
    driven at 3GPP that supports conventional and DSR optimized codecs=20
    and possible exchange of speech meta-information (e.g. data that=20
    may be required to facilitate and enhance the server-side=20
    processing of the input speech and facilitate the dialog management=20
    in an automated voice service. These may include keypad events=20
    over-riding spoken input, notification that the UE is in hands-free=20
    mode, client-side collected information (speech/no-speech, barge-
    in), etc=85.).=20
    - SOAP over SIP or BEEP to support the framework described in=20
    section 1 can also support VCR controls.=20
    - real-time messaging between engine and control is supported=20
    within the framework (e.g. via SOAP or XML events). The framework=20
    also support exchange between engines (same process; see also WSXL=20
    [WS7]).=20
    =20
    Although non-normative, the web service framework described=20
    probably deserves marks of P+ to T.=20
 =20
=0A=
         8.8. Analysis of Security considerations=20
=0A=
           8.8.1. Security Considerations [11]=20
    Web services are evolving to provide security, authentication,=20
    encryption, trust management and privacy . Details can be found for=20
    example in [WS9] and explained in [WS10]. This is now an OASIS=20
    activity [WS11].=20
    =20
    This framework would enable SPEECHSC to employ the security=20
    mechanism provided bu WS-Security for the remote control aspects.=20
    Exchanged media can rely on security mechanism at the transport /=20
    streaming level.=20
    =20
    The web service framework described probably deserves marks of P+=20
    to T.=20
    =20
        =20
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      9. =0D         Security Considerations =20
     =20
    Security considerations for the SPEECHSC protocol are covered by=20
    the comparison against the specific Security requirements in the=20
    SPEECHSC requirements document [1]. =20
        =20
      10. References =20
    [1] E. Burger, =93Requirements for Distributed Control of ASR, SR =
and=20
    TTS Resources" draft-ietf-speechsc-reqts-00.txt, July 31, 2002.=20
    =20
    [2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.=20
    Peterson, R. Sparks, M. Handley, E.Schooler, SIP: Session=20
    Initiation Protocol, RFC3265, June 2002. (Obsoletes RFC2543)=20
    =20
    [WS1] W3C Web Services, http://www.w3c.org/2002/ws/=20
    [WS2] Simple Object Access Protocol (SOAP)=20
    http://www.w3c.org/2002/ws/=20
    [WS3] Web Services Description Language (WSDL 1.1), W3C Note 15=20
    March =20
        2001, http://www.w3.org/TR/wsdl.=20
    [WS4] Leymann, F., Web Service Flow Language, WSFL 1.0, May 2001, =20
         http://www-
    4.ibm.com/software/solutions/webservices/pdf/WSFL.pdf=20
    [WS5] UDDI, http://www.uddi.org/specification.html =20
    [WS6] W3C Voice Activity, http://www.w3c.org/Voice/ =20
    [WS7] WSXL - Web Service eXperience Language submitted to OASIS=20
    WSIA =20
          and WSRP - WSXL - Web Service eXperience Language submitted=20
    to =20
          OASIS WSIA and WSRP=20
    [WS8] Requirements for Distributed Control of ASR, SI/SV and TTS=20
    Resources, =20
    draft-ietf-speechsc-reqts-01.txt=20
    [WS9] Security in a Web Services World: A Proposed Architecture and=20
    Roadmap, =20
    April 7, 2002, Version 1.0, http://www.verisign.com/wss/wss.pdf=20
    [WS10] Kapil Apshankar, WS-Security, Security for Web Services,=20
    http://www.webservicesarchitect.com/content/articles/apshankar04.as
    p=20
    [WS11] OASIS Web Services Security TC, http://www.oasis-
    open.org/committees/wss/=20
    =20
 Author=92s Address=20
        =20
    Brian Wyld =20
    Eloquant SA=20
    ZA Malvaisin             Phone:  +33 476 77 46 92 =20
    Le Versoud, France             Email:  brian.wyld@eloquant.com =20
     =20
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 =20
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 Wyld                    Expires =96 April 2003              [Page 27] =
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                 SPEECHSC Protocol Evaluation Template  November 2002=20
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    are included on all such copies and derivative works.  However,=20
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Subject: Re: [Speechsc] Updated version (v0.2) for speechsc WG Internet Draft 
 submission protocol evaluation document
References: <003501c27e11$3b6d5540$8300010a@hq.eloquant.com>
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some comments/questions on the document.

Comment 1:
My evaluations of MRCP were for the most part, how the protocol already met the
requirements.  And hence when it came to dealing with issues like SI/SV, they were
classified as not supported.

After reading some of the evaluations(such as web services), which pretty much does not
directly  meet most of the requirements. But seems to discuss mostly about how it can
evolve with the additon of  a NEW set of messaging to meet the requirements.

If this is indeed to basis of protocol evaluation and considering MRCP was originally
designed to be extensible for these exact needs.  Even preliminary methods and formats
for these resources were defined but not refined/implemented/tested due to priorities.
The MRCP protocol evaluation would be (T) in most of the SI/SV requirements.

Other comments:
Based on the above, the following sections will change in meeting the requirments.
In all the following sections, texts already provided, addresses how MRCP could be
evolved to meet the needs in FULL, if it does' not already meet it in full.

7.2.4. Protocol efficiency [5.4]  - would become T

7.2.8. Multiple media sessions [5.8]  - would become T

7.4.3. Grammar Specification [7.3.1] - would become T

7.4.4. Explicit Indication of Grammar Format [7.3.2]  - should be T already without any
needed changes.

7.4.5. Grammar Sharing [7.3.3] - This again would be T. since the DEFINE-GRAMMAR method
could be extended to define grammars that cna be shared across sessions, and URI
mechanism used to identify, define and refer to such grammars.

7.5. Analysis of Speaker Identification and Verification Requirements - This whole
section would then be T, since this can be achieved by additon new methods to address
these needs, and since MRCP was designed with these future extensions in mind.

7.6.3. Multiple services in parallel [9.1.2]  - T if 7.5 is extensions are added.

thx,
Sarvi



Brian Wyld wrote:

> Hi,
>
> Sorry, but attached is an UPDATE to the draft I just submitted to take
> account of last minute additions. This draft is v0.2. I have also re-created
> the .txt to be in US Letter size format as per most IETF docs....
>
> I hope this version can be the legitimate Internet Draft for speechsc
> protocol evaluation. Please ignore the previous version.
>
> Thanks
>
> Brian
>
> [Brian Wyld] [brian.wyld@eloquant.com]
> [Directeur General R&D]
> [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
> [advanced solutions for telecoms and IT services]
>
>   ------------------------------------------------------------------------
>                                            Name: draft-speechsc-protocol-eval-v02.txt
>    draft-speechsc-protocol-eval-v02.txt    Type: Plain Text (text/plain)
>                                        Encoding: quoted-printable

--------------3F481D11D43508A4B8BB0868
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begin:vcard 
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adr:;;;;;;
version:2.1
email;internet:sarvi@cisco.com
title:Technical Leader
fn:Sarvi Shanmugham
end:vcard

--------------3F481D11D43508A4B8BB0868--

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From mailnull@www1.ietf.org  Mon Nov  4 15:39:11 2002
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Subject: Re: [Speechsc] Updated version (v0.2) for speechsc WG Internet Draft  submission protocol evaluation document
From: "Stephane H. Maes" <stephane.maes@oracle.com>
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Sarvi,

I will comment in more details later.

I believe that different protocol proposals have to be considered 
differently whne mapped on the requirements.

The web service proposal is at a different level than Sip or Beep for 
example to which it could be bound.

Such fuzziness of the boundary may be frustrating; but it is inherent to 
the nature of SPEECHSC that is also fuzzy between low and more app level 
(engine APIs).

As such, web services have indeed this inherent capability to provide 
any API, support combination and composition etc and to be essentially 
bound to any underlying transport.

Yes syntax is to be defined; but that remains the only major work that 
needs to be done ...
All this remains within the confines of the web service framework.  Can 
you do the same with SIP or BEEP? Yes; but you need to produce a new 
release of the spec or carry lots of info with mechaniss designed only 
for limited extensions.

And by the way, of course web service have no sythax to control speech 
engines. So we could have given failed across the board. But that's not 
the point of the excercise; at least I hope so... Web services is a 
framework that can be applied unchanged (+/-) to support speechsc... And 
yes syntax has to be defined. Honnestky that' comon practice and the 
work of a few months at the level of a standard working group.

With MRCP, I do no know how to position it either. It is a syntax on top 
of another protocol. As defined it does not meet the requirements. But I 
understand that the argument that maybe only syntax needs to be defined 
to satisfy this. So you are rigth to make these updates as MRCP is 
anyway a draft and can therefore support that.

Again the evaluation based on the requirementas is may not yet be enough 
to decide and pick an approach. Other design points may have to be taken 
into account to progress.

I agree with you that whatever is the point of view taken; it is 
frustrating when considered from another angle. We will have to decide 
what would best help progress.

Thanks

Stephane

On Mon, 4 Nov 2002 3:03PM -0500, Sarvi Shanmugham wrote:
> some comments/questions on the document.
>
> Comment 1:
> My evaluations of MRCP were for the most part, how the protocol already 
> met the
> requirements.  And hence when it came to dealing with issues like 
> SI/SV, they were
> classified as not supported.
>
> After reading some of the evaluations(such as web services), which 
> pretty much does not
> directly  meet most of the requirements. But seems to discuss mostly 
> about how it can
> evolve with the additon of  a NEW set of messaging to meet the 
> requirements.
>
> If this is indeed to basis of protocol evaluation and considering MRCP 
> was originally
> designed to be extensible for these exact needs.  Even preliminary 
> methods and formats
> for these resources were defined but not refined/implemented/tested due 
> to priorities.
> The MRCP protocol evaluation would be (T) in most of the SI/SV 
> requirements.
>
> Other comments:
> Based on the above, the following sections will change in meeting the 
> requirments.
> In all the following sections, texts already provided, addresses how 
> MRCP could be
> evolved to meet the needs in FULL, if it does' not already meet it in 
> full.
>
> 7.2.4. Protocol efficiency [5.4]  - would become T
>
> 7.2.8. Multiple media sessions [5.8]  - would become T
>
> 7.4.3. Grammar Specification [7.3.1] - would become T
>
> 7.4.4. Explicit Indication of Grammar Format [7.3.2]  - should be T 
> already without any
> needed changes.
>
> 7.4.5. Grammar Sharing [7.3.3] - This again would be T. since the 
> DEFINE-GRAMMAR method
> could be extended to define grammars that cna be shared across 
> sessions, and URI
> mechanism used to identify, define and refer to such grammars.
>
> 7.5. Analysis of Speaker Identification and Verification Requirements - 
> This whole
> section would then be T, since this can be achieved by additon new 
> methods to address
> these needs, and since MRCP was designed with these future extensions 
> in mind.
>
> 7.6.3. Multiple services in parallel [9.1.2]  - T if 7.5 is extensions 
> are added.
>
> thx,
> Sarvi
>
>
>
> Brian Wyld wrote:
>
>>  Hi,
>>
>>  Sorry, but attached is an UPDATE to the draft I just submitted to 
>> take
>>  account of last minute additions. This draft is v0.2. I have also 
>> re-created
>>  the .txt to be in US Letter size format as per most IETF docs....
>>
>>  I hope this version can be the legitimate Internet Draft for speechsc
>>  protocol evaluation. Please ignore the previous version.
>>
>>  Thanks
>>
>>  Brian
>>
>>  [Brian Wyld] [brian.wyld@eloquant.com]
>>  [Directeur General R&D]
>>  [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
>>  [advanced solutions for telecoms and IT services]
>>
>>    
>> ------------------------------------------------------------------------
>>                                             Name: 
>> draft-speechsc-protocol-eval-v02.txt
>>     draft-speechsc-protocol-eval-v02.txt    Type: Plain Text 
>> (text/plain)
>>                                         Encoding: quoted-printable

___
Stephane H. Maes, PhD,
Director of Architecture, Mobile, Oracle Corporation.
Ph: +1-203-300-7786 (mobile/SMS); Fax: +1-203-798-6948
e-mail: stephane.maes@oracle.com
IM: shmaes (AIM) or stephane_maes@hotmail.com (MSN)
_______________________________________________
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Date: Mon, 04 Nov 2002 12:55:20 -0800
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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CC: brian.wyld@eloquant.com, "Speechsc (E-mail)" <speechsc@ietf.org>
Subject: Re: [Speechsc] Updated version (v0.2) for speechsc WG Internet Draft  
 submission protocol evaluation document
References: <1036442448.17309658@r5.dngr.org>
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Hi Stephane,
     I was in no way claiming that your analysis of web services is wrong.

But like you mention, since my evaluation of MRCP is based on the notion that
the current MRCP spec will be extended in one of the 3 ways proposed, I think
the following changes are appropriate.

So I was just making these changes to make the evaluation consistent with the
Webservices evalution.

Sarvi

"Stephane H. Maes" wrote:

> Sarvi,
>
> I will comment in more details later.
>
> I believe that different protocol proposals have to be considered
> differently whne mapped on the requirements.
>
> The web service proposal is at a different level than Sip or Beep for
> example to which it could be bound.
>
> Such fuzziness of the boundary may be frustrating; but it is inherent to
> the nature of SPEECHSC that is also fuzzy between low and more app level
> (engine APIs).
>
> As such, web services have indeed this inherent capability to provide
> any API, support combination and composition etc and to be essentially
> bound to any underlying transport.
>
> Yes syntax is to be defined; but that remains the only major work that
> needs to be done ...
> All this remains within the confines of the web service framework.  Can
> you do the same with SIP or BEEP? Yes; but you need to produce a new
> release of the spec or carry lots of info with mechaniss designed only
> for limited extensions.
>
> And by the way, of course web service have no sythax to control speech
> engines. So we could have given failed across the board. But that's not
> the point of the excercise; at least I hope so... Web services is a
> framework that can be applied unchanged (+/-) to support speechsc... And
> yes syntax has to be defined. Honnestky that' comon practice and the
> work of a few months at the level of a standard working group.
>
> With MRCP, I do no know how to position it either. It is a syntax on top
> of another protocol. As defined it does not meet the requirements. But I
> understand that the argument that maybe only syntax needs to be defined
> to satisfy this. So you are rigth to make these updates as MRCP is
> anyway a draft and can therefore support that.
>
> Again the evaluation based on the requirementas is may not yet be enough
> to decide and pick an approach. Other design points may have to be taken
> into account to progress.
>
> I agree with you that whatever is the point of view taken; it is
> frustrating when considered from another angle. We will have to decide
> what would best help progress.
>
> Thanks
>
> Stephane
>
> On Mon, 4 Nov 2002 3:03PM -0500, Sarvi Shanmugham wrote:
> > some comments/questions on the document.
> >
> > Comment 1:
> > My evaluations of MRCP were for the most part, how the protocol already
> > met the
> > requirements.  And hence when it came to dealing with issues like
> > SI/SV, they were
> > classified as not supported.
> >
> > After reading some of the evaluations(such as web services), which
> > pretty much does not
> > directly  meet most of the requirements. But seems to discuss mostly
> > about how it can
> > evolve with the additon of  a NEW set of messaging to meet the
> > requirements.
> >
> > If this is indeed to basis of protocol evaluation and considering MRCP
> > was originally
> > designed to be extensible for these exact needs.  Even preliminary
> > methods and formats
> > for these resources were defined but not refined/implemented/tested due
> > to priorities.
> > The MRCP protocol evaluation would be (T) in most of the SI/SV
> > requirements.
> >
> > Other comments:
> > Based on the above, the following sections will change in meeting the
> > requirments.
> > In all the following sections, texts already provided, addresses how
> > MRCP could be
> > evolved to meet the needs in FULL, if it does' not already meet it in
> > full.
> >
> > 7.2.4. Protocol efficiency [5.4]  - would become T
> >
> > 7.2.8. Multiple media sessions [5.8]  - would become T
> >
> > 7.4.3. Grammar Specification [7.3.1] - would become T
> >
> > 7.4.4. Explicit Indication of Grammar Format [7.3.2]  - should be T
> > already without any
> > needed changes.
> >
> > 7.4.5. Grammar Sharing [7.3.3] - This again would be T. since the
> > DEFINE-GRAMMAR method
> > could be extended to define grammars that cna be shared across
> > sessions, and URI
> > mechanism used to identify, define and refer to such grammars.
> >
> > 7.5. Analysis of Speaker Identification and Verification Requirements -
> > This whole
> > section would then be T, since this can be achieved by additon new
> > methods to address
> > these needs, and since MRCP was designed with these future extensions
> > in mind.
> >
> > 7.6.3. Multiple services in parallel [9.1.2]  - T if 7.5 is extensions
> > are added.
> >
> > thx,
> > Sarvi
> >
> >
> >
> > Brian Wyld wrote:
> >
> >>  Hi,
> >>
> >>  Sorry, but attached is an UPDATE to the draft I just submitted to
> >> take
> >>  account of last minute additions. This draft is v0.2. I have also
> >> re-created
> >>  the .txt to be in US Letter size format as per most IETF docs....
> >>
> >>  I hope this version can be the legitimate Internet Draft for speechsc
> >>  protocol evaluation. Please ignore the previous version.
> >>
> >>  Thanks
> >>
> >>  Brian
> >>
> >>  [Brian Wyld] [brian.wyld@eloquant.com]
> >>  [Directeur General R&D]
> >>  [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
> >>  [advanced solutions for telecoms and IT services]
> >>
> >>
> >> ------------------------------------------------------------------------
> >>                                             Name:
> >> draft-speechsc-protocol-eval-v02.txt
> >>     draft-speechsc-protocol-eval-v02.txt    Type: Plain Text
> >> (text/plain)
> >>                                         Encoding: quoted-printable
>
> ___
> Stephane H. Maes, PhD,
> Director of Architecture, Mobile, Oracle Corporation.
> Ph: +1-203-300-7786 (mobile/SMS); Fax: +1-203-798-6948
> e-mail: stephane.maes@oracle.com
> IM: shmaes (AIM) or stephane_maes@hotmail.com (MSN)

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fn:Sarvi Shanmugham
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--------------57243E34E333931866613885--

_______________________________________________
Speechsc mailing list
Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



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Subject: Re: [Speechsc] Updated version (v0.2) for speechsc WG Internet Draft   submission protocol evaluation document
From: "Stephane H. Maes" <stephane.maes@oracle.com>
References: <1036442448.17309658@r5.dngr.org> <3DC6DEB7.9931CE30@cisco.com>
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I did not assume that you were... Don't worry. I think that you pointed 
out a challenge in using this to make any type of progress.

Thanks

Stephane

On Mon, 4 Nov 2002 3:56PM -0500, Sarvi Shanmugham wrote:
> Hi Stephane,
>      I was in no way claiming that your analysis of web services is 
> wrong.
>
> But like you mention, since my evaluation of MRCP is based on the 
> notion that
> the current MRCP spec will be extended in one of the 3 ways proposed, I 
> think
> the following changes are appropriate.
>
> So I was just making these changes to make the evaluation consistent 
> with the
> Webservices evalution.
>
> Sarvi
>
> "Stephane H. Maes" wrote:
>
>>  Sarvi,
>>
>>  I will comment in more details later.
>>
>>  I believe that different protocol proposals have to be considered
>>  differently whne mapped on the requirements.
>>
>>  The web service proposal is at a different level than Sip or Beep for
>>  example to which it could be bound.
>>
>>  Such fuzziness of the boundary may be frustrating; but it is inherent 
>> to
>>  the nature of SPEECHSC that is also fuzzy between low and more app 
>> level
>>  (engine APIs).
>>
>>  As such, web services have indeed this inherent capability to provide
>>  any API, support combination and composition etc and to be 
>> essentially
>>  bound to any underlying transport.
>>
>>  Yes syntax is to be defined; but that remains the only major work 
>> that
>>  needs to be done ...
>>  All this remains within the confines of the web service framework.  
>> Can
>>  you do the same with SIP or BEEP? Yes; but you need to produce a new
>>  release of the spec or carry lots of info with mechaniss designed 
>> only
>>  for limited extensions.
>>
>>  And by the way, of course web service have no sythax to control 
>> speech
>>  engines. So we could have given failed across the board. But that's 
>> not
>>  the point of the excercise; at least I hope so... Web services is a
>>  framework that can be applied unchanged (+/-) to support speechsc... 
>> And
>>  yes syntax has to be defined. Honnestky that' comon practice and the
>>  work of a few months at the level of a standard working group.
>>
>>  With MRCP, I do no know how to position it either. It is a syntax on 
>> top
>>  of another protocol. As defined it does not meet the requirements. 
>> But I
>>  understand that the argument that maybe only syntax needs to be 
>> defined
>>  to satisfy this. So you are rigth to make these updates as MRCP is
>>  anyway a draft and can therefore support that.
>>
>>  Again the evaluation based on the requirementas is may not yet be 
>> enough
>>  to decide and pick an approach. Other design points may have to be 
>> taken
>>  into account to progress.
>>
>>  I agree with you that whatever is the point of view taken; it is
>>  frustrating when considered from another angle. We will have to 
>> decide
>>  what would best help progress.
>>
>>  Thanks
>>
>>  Stephane
>>
>>  On Mon, 4 Nov 2002 3:03PM -0500, Sarvi Shanmugham wrote:
>>  > some comments/questions on the document.
>>  >
>>  > Comment 1:
>>  > My evaluations of MRCP were for the most part, how the protocol 
>> already
>>  > met the
>>  > requirements.  And hence when it came to dealing with issues like
>>  > SI/SV, they were
>>  > classified as not supported.
>>  >
>>  > After reading some of the evaluations(such as web services), which
>>  > pretty much does not
>>  > directly  meet most of the requirements. But seems to discuss 
>> mostly
>>  > about how it can
>>  > evolve with the additon of  a NEW set of messaging to meet the
>>  > requirements.
>>  >
>>  > If this is indeed to basis of protocol evaluation and considering 
>> MRCP
>>  > was originally
>>  > designed to be extensible for these exact needs.  Even preliminary
>>  > methods and formats
>>  > for these resources were defined but not refined/implemented/tested 
>> due
>>  > to priorities.
>>  > The MRCP protocol evaluation would be (T) in most of the SI/SV
>>  > requirements.
>>  >
>>  > Other comments:
>>  > Based on the above, the following sections will change in meeting 
>> the
>>  > requirments.
>>  > In all the following sections, texts already provided, addresses 
>> how
>>  > MRCP could be
>>  > evolved to meet the needs in FULL, if it does' not already meet it 
>> in
>>  > full.
>>  >
>>  > 7.2.4. Protocol efficiency [5.4]  - would become T
>>  >
>>  > 7.2.8. Multiple media sessions [5.8]  - would become T
>>  >
>>  > 7.4.3. Grammar Specification [7.3.1] - would become T
>>  >
>>  > 7.4.4. Explicit Indication of Grammar Format [7.3.2]  - should be T
>>  > already without any
>>  > needed changes.
>>  >
>>  > 7.4.5. Grammar Sharing [7.3.3] - This again would be T. since the
>>  > DEFINE-GRAMMAR method
>>  > could be extended to define grammars that cna be shared across
>>  > sessions, and URI
>>  > mechanism used to identify, define and refer to such grammars.
>>  >
>>  > 7.5. Analysis of Speaker Identification and Verification 
>> Requirements -
>>  > This whole
>>  > section would then be T, since this can be achieved by additon new
>>  > methods to address
>>  > these needs, and since MRCP was designed with these future 
>> extensions
>>  > in mind.
>>  >
>>  > 7.6.3. Multiple services in parallel [9.1.2]  - T if 7.5 is 
>> extensions
>>  > are added.
>>  >
>>  > thx,
>>  > Sarvi
>>  >
>>  >
>>  >
>>  > Brian Wyld wrote:
>>  >
>>  >>  Hi,
>>  >>
>>  >>  Sorry, but attached is an UPDATE to the draft I just submitted to
>>  >> take
>>  >>  account of last minute additions. This draft is v0.2. I have also
>>  >> re-created
>>  >>  the .txt to be in US Letter size format as per most IETF docs....
>>  >>
>>  >>  I hope this version can be the legitimate Internet Draft for 
>> speechsc
>>  >>  protocol evaluation. Please ignore the previous version.
>>  >>
>>  >>  Thanks
>>  >>
>>  >>  Brian
>>  >>
>>  >>  [Brian Wyld] [brian.wyld@eloquant.com]
>>  >>  [Directeur General R&D]
>>  >>  [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
>>  >>  [advanced solutions for telecoms and IT services]
>>  >>
>>  >>
>>  >> 
>> ------------------------------------------------------------------------
>>  >>                                             Name:
>>  >> draft-speechsc-protocol-eval-v02.txt
>>  >>     draft-speechsc-protocol-eval-v02.txt    Type: Plain Text
>>  >> (text/plain)
>>  >>                                         Encoding: quoted-printable
>>
>>  ___
>>  Stephane H. Maes, PhD,
>>  Director of Architecture, Mobile, Oracle Corporation.
>>  Ph: +1-203-300-7786 (mobile/SMS); Fax: +1-203-798-6948
>>  e-mail: stephane.maes@oracle.com
>>  IM: shmaes (AIM) or stephane_maes@hotmail.com (MSN)

___
Stephane H. Maes, PhD,
Director of Architecture, Mobile, Oracle Corporation.
Ph: +1-203-300-7786 (mobile/SMS); Fax: +1-203-798-6948
e-mail: stephane.maes@oracle.com
IM: shmaes (AIM) or stephane_maes@hotmail.com (MSN)
_______________________________________________
Speechsc mailing list
Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



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Excellent news from Harald!  This is an excellent tool; use it if you are wondering where your draft is.

-----Original Message-----
From: Harald Tveit Alvestrand [mailto:harald@alvestrand.no]
Sent: Monday, November 04, 2002 4:44 PM
Subject: Introducing the ID tracker



Hello,

The IESG is always trying to work better.
We have gotten a number of comments from the community that it is sometimes 
very hard to find out what the IESG is doing to documents, why documents 
take a long time to process, who makes comments, what the comments are and 
so on, and that this is a significant source of frustration for the 
community.

For the last year or so, the secretariat has been working on a tool to help 
us keep track of what documents are on our plate, what state they are in 
and who is responsible for them; we call it the "ID tracker". It's been in 
use for about 6 months now, and has definitely improved our ability to keep 
track.

For several months, the "status of items" pages on the IESG web pages have 
been generated from this tool, showing the high-order bit of who we think 
is responsible.

In order to help you know more about what we are doing, we've decided to 
open up a public view of the ID tracker itself.

This will allow you to search for the documents you want to look at, and 
for each document see:

- What its current state is
- What the history of IESG processing is
- AD and IESG comments on the document
- For documents in IESG consideration for standards track, what the 
individual ADs have indicated as their opinion on the document, and if they 
think there are problems with it, what the comments are.

The last point is a revision of previous IESG policy, based on the feedback 
from the Yokohama plenary; we are now telling you the names that go with 
each comment from the IESG members.

The system is far from perfect - you can tell from the history of documents 
that we've been changing this as we go along - but we hope that it will be 
a useful tool for people to figure out what the IESG is doing with their 
documents, and for them to have an easier time talking to the relevant ADs 
in order to get what we all want out of the process - relevant, high 
quality standards for the Internet.

The tool and its documentation is found at

https://datatracker.ietf.org/public/pidtracker.cgi

The link to it is also visible on the IESG Web page.

Welcome!

              The IESG

_______________________________________________
Speechsc mailing list
Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Speech Services Control Working Group of the IETF.

	Title		: SPEECHSC Protocol Evaluation
	Author(s)	: B. Wyld
	Filename	: draft-ietf-speechsc-protocol-eval-01.txt
	Pages		: 28
	Date		: 2002-11-5
	
This document is the Protocol Evaluation Document for the SPEECHSC 
Working Group.  Section 3 provides the summary of the  individual 
protocol comparisons (in the sections 4-N following) against the 
SPEECHSC requirements [1].

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-protocol-eval-01.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-speechsc-protocol-eval-01.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-speechsc-protocol-eval-01.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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_______________________________________________
Speechsc mailing list
Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



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Thread-Topic: Draft Agenda for speechsc
Thread-Index: AcKFoWP+Tu30HN79SI+o5aQsHo+mLAACU+WH
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To: <agenda@ietf.org>
Cc: <speechsc@ietf.org>
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Charter: http://www.ietf.org/html.charters/speechsc-charter.html <http://www.ietf.org/html.charters/speechsc-charter.html> 
List Archive: http://www1.ietf.org/mail-archive/working-groups/speechsc/current/maillist.html <http://www1.ietf.org/mail-archive/working-groups/speechsc/current/maillist.html> 
Supplemental page: http://flyingfox.snowshore.com/i-d <http://flyingfox.snowshore.com/i-d> 
 
THURSDAY, November 21, 2002
1530-1730 Afternoon Sessions II
Consulate	 TSV	 speechsc	 Speech Services Control WG

 
3:30    5 min      Agenda Bashing
3:35   10 min      Requirements Document Status (chairs)
3:45   50 min      Protocol Analysis Document Open Issues
                          Beep (Jerry Carter)
                          SIP (Rajiv Dharmadhikari)
                          RTSP (Brian Wyld) [won't be in Atlanta]
                          MRCP (Sarvi Shanmugham)
                          Web Services (Stephane Maes)
4:35   30 min      Next Steps, Design Team Fomation
5:05   25 min      The Internet Standards Process and speechsc

 
You are expected to have read the following late night reading material:
Requirements Document: 
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-reqts-02.txt <http://www.ietf.org/internet-drafts/draft-ietf-speechsc-reqts-02.txt> 
Protocol Evaluation Document: 
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-protocol-eval-01.txt <http://www.ietf.org/internet-drafts/draft-ietf-speechsc-protocol-eval-01.txt> 
 
Internet Standards Process:
Instructions to RFC Authors: 
http://www.ietf.org/rfc/rfc2223.txt?number=2223 <http://www.ietf.org/rfc/rfc2223.txt?number=2223> 
New Instructions to RFC Authors (draft): 
http://www.ietf.org/internet-drafts/draft-rfc-editor-rfc2223bis-03.txt <http://www.ietf.org/internet-drafts/draft-rfc-editor-rfc2223bis-03.txt> 
The Internet Standards Process -- Revision 3:
http://www.ietf.org/rfc/rfc2026.txt?number=2026 <http://www.ietf.org/rfc/rfc2026.txt?number=2026> 

_______________________________________________
Speechsc mailing list
Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



From mailnull@www1.ietf.org  Thu Nov  7 15:46:18 2002
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To: All IETF Working Groups: ;
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Subject: [Speechsc] Note Well Statement
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From time to time, especially just before a meeting, this statement is to
be sent to each and every IETF working group mailing list.
===========================================================================

				NOTE WELL

All statements related to the activities of the IETF and addressed to the
IETF are subject to all provisions of Section 10 of RFC 2026, which grants
to the IETF and its participants certain licenses and rights in such
statements.

Such statements include verbal statements in IETF meetings, as well as
written and electronic communications made at any time or place, which are
addressed to

    - the IETF plenary session,
    - any IETF working group or portion thereof,
    - the IESG, or any member thereof on behalf of the IESG,
    - the IAB or any member thereof on behalf of the IAB,
    - any IETF mailing list, including the IETF list itself,
      any working group or design team list, or any other list
      functioning under IETF auspices,
    - the RFC Editor or the Internet-Drafts function

Statements made outside of an IETF meeting, mailing list or other function,
that are clearly not intended to be input to an IETF activity, group or
function, are not subject to these provisions.
_______________________________________________
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https://www1.ietf.org/mailman/listinfo/speechsc



From mailnull@www1.ietf.org  Mon Nov 18 15:40:44 2002
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Thread-Topic: Homework due Thursday
thread-index: AcKPFzRT6kBkaxV4RGuNuKN6lYCy6A==
From: "Eric Burger" <eburger@snowshore.com>
To: "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
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This is a reminder for people new to the IETF.  Please disregard if this is not new to you.

Presentations at the Work Group meeting revolve around open issues.  We have only a limited amount of time.  This means that we cannot have presentations of a tutorial nature (e.g., "this is what my draft is about" or "here are the features of protocol X").  Rather, the presentations address open issues (e.g., "this point had dissent on the list, here is why the draft is right / wrong").


Please be prepared for the meeting.  Review the drafts:

Requirements Document: 
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-reqts-02.txt

Protocol Evaluation Document: 
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-protocol-eval-01.txt

Also, it may be beneficial to review the discussion on the list:
http://www1.ietf.org/mail-archive/working-groups/speechsc/current/maillist.html



If you read and understand the following documents, you can leave the work group meeting early.  However, if you are an "old hand", we would encourage you to stick around and give the new folks guidance.

New Instructions to RFC Authors (draft): 
http://www.ietf.org/internet-drafts/draft-rfc-editor-rfc2223bis-03.txt

The Internet Standards Process -- Revision 3:
http://www.ietf.org/rfc/rfc2026.txt?number=2026
_______________________________________________
Speechsc mailing list
Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



From mailnull@www1.ietf.org  Mon Nov 18 16:26:53 2002
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Thread-Topic: Scribe and Transcriber Call for Participation
thread-index: AcKPFA4u8WueWqjtTWi71mtQUb+Z3A==
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We need two volunteers for our meeting on Thursday.

The first is a scribe.  We would be happy to have two or three volunteers.  We can split the load so you don't have to take notes for the entire meeting.  The agenda is on line at http://www.ietf.org/ietf/02nov/speechsc.txt .  If you are presenting open issues, you are probably NOT a candidate for being scribe.

The second is a transcriber.  All IETF WG sessions have a parallel Jabber (IRC) session.  We need someone to write out what is going on in the session, as well as take questions from the Internet audience.  You will need 802.11 access from your computer for this job.  Again, we can take more than one person for this job.  If you are presenting open issues, you are not a candidate for being transcriber.
_______________________________________________
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Speechsc@ietf.org
https://www1.ietf.org/mailman/listinfo/speechsc



From mailnull@www1.ietf.org  Tue Nov 19 04:48:09 2002
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Thread-Topic: text conferencing at the 55th IETF meeting in Atlanta
thread-index: AcKPDa1H3P9lrRA4SEuvBXfm/CDLZAAlbrOw
From: "Eric Burger" <eburger@snowshore.com>
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For the remote, Internet audience, we have two choices for a live text feed of the proceedings.  The first is Jabber and the second is SIP.  Note the SIP implementation is an Alpha version.  I have no idea if it works.  If either tool is broken, contact jabber.com or iptel.org, respectively.

We will not have MBONE access or an audio conference bridge.

For Jabber, the conference room is "speechsc" at conference.ietf.jabber.com .
The recording is at http://www.jabber.com/chatbot/logs/conference.ietf.jabber.com/speechsc/ .

For SIP, check out http://www.iptel.org/ietf55/ .  Our conference is speechsc.

As a reminder, out meeting is in the Consulate room from 3:30pm - 5:30pm ET on Thursday November 21.


-----Original Message-----
From: Marshall Rose [mailto:mrose+mtr.netnews@dbc.mtview.ca.us]
Sent: Monday, November 18, 2002 9:11 AM
To: ietf@ietf.org
Subject: text conferencing at the 55th IETF meeting in Atlanta


	     Remote Access for the 55th IETF meeting in Atlanta:
			     Text Conferencing

At each IETF meeting, two of the working group meeting rooms are equipped
for video multicast and remote participation.  That is, for every IETF
meeting slot, two of the working groups can see and hear the
meeting. For the 55th IETF, in *addition* to the usual network A/V, text
conferencing will be provided for every working group that meets.

All of the conference rooms are hosted on

    conference.ietf.jabber.com

and each is named using the official IETF abbreviation found in the
agenda (e.g., "apparea",  "dhc", "forces", and so on -- for all the
examples that follow, we'll use "foobar" as the abbreviation).

Each conference room also has a 'bot which records everything that gets
sent:
    
    http://www.jabber.com/chatbot/logs/conference.ietf.jabber.com/foobar/

Enjoy!
    
/mtr

-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Monday, November 18, 2002 10:33 AM
To: Marshall Rose; ietf@ietf.org
Cc: serhelp@iptel.org
Subject: Re: text conferencing at the 55th IETF meeting in Atlanta


For those, who   would like to use SIP for text conferencing,
experimental operation of a SIP2Jabber gateway has been set up at
iptel.org. See http://www.iptel.org/ietf55/ for guidelines how
to join the IETF chat rooms.

Better set your expectations low  -- the gateway has not been tested 
with bigger user populations until now.

-Jiri
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Posted to http://flyingfox.snowshore.com/i-d/speechsc/

Sorry, we only have PowerPoint right now.  PostScript will come after the meeting.
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The following slides are empty:

Chairs Slides  

MRCP Analysis Slides 

Web Services Analysis Slides 

Internet Draft Tutorial, or I-D 101 


-- Qiru

Eric Burger wrote:
> 
> Posted to http://flyingfox.snowshore.com/i-d/speechsc/
> 
> Sorry, we only have PowerPoint right now.  PostScript will come after the meeting.
> _______________________________________________
> Speechsc mailing list
> Speechsc@ietf.org
> https://www1.ietf.org/mailman/listinfo/speechsc
_______________________________________________
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From: Edward Epstein <eae@us.ibm.com>
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I will be out of the office starting November 22, 2002 and will not return
until December 2, 2002.

I will respond to your message when I return.

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From: "Neil Deason" <ndeason@ubiquity.net>
To: <speechsc@ietf.org>
Date: Sun, 24 Nov 2002 12:11:10 -0800
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Subject: [Speechsc] SPEECHSC & usage of SIP
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Hi.

I would like to offer a perspective on the use of SIP within the problem
space being studied by the group. SIP is a protocol for initiating,
modifying and terminating sessions. SIP is therefore not best suited as
a general transport protocol for other protocols or a media resource
control protocol.

With that said the most appropriate use for SIP in this context would
appear to be to set up the control session, which then runs the TBD
SPEECHSC protocol. This has some attractive properties - it provides a
natural association between the audio and control sessions and SIP is
likely to be widely deployed in the networks where the SPEECHSC protocol
is needed.

At the WG meeting I mentioned a now expired Internet Draft which
provided an example of a similar usage of SIP to set up SOAP sessions.
This is not intended to say SOAP must be the SPEECHSC protocol. Though
the thought of MRCP encoded in SOAP and set up through SIP might have
some appeal here ;)

Some important points to note about this draft - it has expired and is
out of date. In particular the SIMPLE WG appears to have moved towards
CPIM/TCP in preference to IMTP for message based sessions where SIP is
concerned. Therefore the idea in this draft for using IMTP as the
transport for SOAP messaging needs revision. However, the example usage
part of how an audio/RTP session can be set up along side a control
channel through SIP/SDP is still valid.

http://www.ietf.org/internet-drafts/draft-deason-sipping-soap-sessions-0
0.txt

hth,
Neil.
--
Neil Deason                             3 Lagoon Drive
ndeason@ubiquity.net                    Suite 345
Director of Technology                  Redwood City
Ubiquity Software Corporation           CA 94065
http://www.ubiquity.net                 USA

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