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Date: Wed, 1 Oct 2003 15:48:39 -0400
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Subject: [Speechsc] Preliminary Meeting Slot
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THURSDAY, November 13, 2003
1300-1500 Afternoon Sessions I
TSV speechsc Speech Services Control WG=20


Subject to change without notice.


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Subject: [Speechsc] MRCPv2  comments
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Hi,

These are some things that I noticed on a quick inspection of MRCP v2.0  =
spec.

Fetching, Storing and deleting TTS content
-------------------------------------------------------------

1. When the TTS receives an inline content, it has a content-id =
associated with it. If it has a cache-control field, then that would =
determine how long this content is kept on the server.  However in case =
of a scenario where there MIGHT be a need to keep TTS content for an =
entire page browse (so no cache control is activated) and then if the =
user navigates to a new page, the client would decide to delete that =
particular TTS content, it cannot do so. Basically, there is no =
DELETE_CONTENT directive that can take the "content-id" as a parameter. =
If there are other ways to do this, these are not so clear.=20


2. For TTS content, the spec doesnt specify if content-id is needed for =
both inline content and URI's. It specifies for ASR that content-id is =
needed only for inline recognition grammars and not for external ones.

3. The spec does not specify which all TTS methods can have the content =
as a payload. I would assume from the spec that only the SPEAK request =
would have the content as a payload. However, the spec defines message =
headers like "audio fetch hint" and "fetch hint" that is used with =
SET-PARAMS, GET-PARAMS and SPEAK. They have values of "prefetch" =
indicating that the content may be prefetched and "safe" that says the =
content need to be downloaded only when needed. If (according to my =
understanding) there is prefetch enabled with a SET-PARAM request, and =
the content can be specified ONLY through a SPEAK request, I dont see a =
prefetch capability here. There should be some method like LOAD-CONTENT =
that can take either an inline content or a URI to fetch (and the =
content addressed through a content-id).=20

Queueing
-------------

MRCPv2 does not specify using a QUEUE method. If there are preloaded =
content (for TTS) on the server that could be referenced using a =
content-id, the client should be able to choose multiple content through =
the id's and create a queue to be played back. It can be argued that =
multiple SPEAK requests can be sent and they naturally form a queue to =
be played back. However, consider the scenario where you have sent =
multiple SPEAK and the TTS is playing them back. Then the client decides =
to PAUSE this and play another TTS content. With the current spec, if =
you pause this, the next SPEAK will be appended to this queue rather =
than spoken immediately. If a STOP gets sent, the whole queue gets =
cleared.=20

If a QUEUE is implemented, then this queue can be paused (through a =
PAUSE_QUEUE or something) and a new TTS can be played. The TTS would =
send back a PLAY_COMPLETE after which the client can resume playback of =
the queue. =20

Audio recorder
---------------------

MRCPv2 does not specify the use of an audio recorder with RECORD, and =
FETCH capability.
There is provision in the ASR methods for recording only those waveforms =
that were used for recognition with fetch capability.

ASR
-------

1. MRCPv2 does not have provision to delete pre-loaded grammars if the =
client decides they are no longer needed.

2. No INTERMEDIATE_RECOGNITION_RESULT send event from the recognizer and =
provision for setting this with the message fields (eg SET-PARAMS)

best regards,

Sailesh.

Sailesh Kumar Sathish      GSM +358 50 4835679
Nokia Research Center      Desk +358 3 272 5679
P.O.Box 100, Tampere      sailesh.sathish@nokia.com
Finland -33721



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From: "Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>
To: "'speechsc@ietf.org'" <speechsc@ietf.org>
Date: Wed, 8 Oct 2003 08:54:01 -0400 
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Subject: [Speechsc] draft-ietf-speechsc-mrcpv2-00: multiple media sessions
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According to Chapter 3.8 of draft-ieft-speechsc-reqts-04 MRCPv2 MUST allow a
1:N mapping between session and RTP channels. But the spec
(draft-ietf-speechsc-mrcpv2-00 chapter 2 last paragraph and chapter 3.3)
says that the media pipes MUST be shared by all media processing resources
under that SIP session. Why this restriction?

How would a sample for the 1:N use case look like? Especially how does the
client know which RTP stream is used for ASR and which for SV/SI (if it does
not have this info it has to send all audio data of every utterance to ASR
and SV/SI, even if only ASR or SV/SI is activated at the same time)? The
"i=" field (of the SDP session description) could be used to label the
different media streams. 

Is the following sequence of messages MRCPv2 compliant?
C->S: 
           INVITE sip:mresources@mediaserver.com SIP/2.0 
           Max-Forwards: 70 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip: sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: 142 
  
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=SDP Seminar 
           i=A session for processing media 
           c=IN IP4 224.2.17.12/127 
           m=control 0 mrcp 01 
           m=control 0 mrcp 02 
           m=control 0 mrcp 04
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=sendrecv 
     
 S->C: 
           SIP/2.0 200 OK 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip: sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: 131 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=SDP Seminar 
           i=A session for processing media 
           c=IN IP4 224.2.17.12/127 
           m=control 32416 mrcp 32AECB23433801 
           m=control 32416 mrcp 32AECB23433802 
           m=control 32416 mrcp 32AECB23433804 
           m=audio 48260 RTP/AVP 0 96
           i=ASR
           c=IN IP4 224.2.17.13/127 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=recvonly 
           m=audio 48260 RTP/AVP 0 96
           i=TTS
           c=IN IP4 224.2.17.14/127 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=sendonly 
           m=audio 48260 RTP/AVP 0 96
           i=SV
           c=IN IP4 224.2.17.15/127 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=recvonly 
     
    C->S: 
           ACK sip:mrcp@mediaserver.com SIP/2.0 
           Max-Forwards: 70 
           To: MediaServer <sip:mrcp@mediaserver.com>;tag=a6c85cf 
           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314164 ACK 
           Content-Length: 0 

And what about the following message sequence?

C->S: 
           INVITE sip:mresources@mediaserver.com SIP/2.0 
           Max-Forwards: 70 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip: sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: 142 
  
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=SDP Seminar 
           i=A session for processing media 
           c=IN IP4 224.2.17.12/127 
           m=control 0 mrcp 01 
           m=control 0 mrcp 02 
           m=control 0 mrcp 04
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=sendrecv 
     
 S->C: 
           SIP/2.0 200 OK 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip: sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: 131 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=SDP Seminar 
           i=A session for processing media 
           m=control 32416 mrcp 32AECB23433801 
           c=IN IP4 224.2.17.13/127 
           m=control 32416 mrcp 32AECB23433802 
           c=IN IP4 224.2.17.14/127 
           m=control 32416 mrcp 32AECB23433804 
           c=IN IP4 224.2.17.15/127 
           m=audio 48260 RTP/AVP 0 96
           i=ASR
           c=IN IP4 224.2.17.13/127 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=recvonly 
           m=audio 48260 RTP/AVP 0 96
           i=TTS 
           c=IN IP4 224.2.17.14/127 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=sendonly 
           m=audio 48260 RTP/AVP 0 96
           i=SV 
           c=IN IP4 224.2.17.15/127 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=recvonly 
     
    C->S: 
           ACK sip:mrcp@mediaserver.com SIP/2.0 
           Max-Forwards: 70 
           To: MediaServer <sip:mrcp@mediaserver.com>;tag=a6c85cf 
           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314164 ACK 
           Content-Length: 0 

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



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Date: Wed, 08 Oct 2003 09:15:22 -0400
From: "David R. Oran" <oran@cisco.com>
To: "Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>,
        "'speechsc@ietf.org'" <speechsc@ietf.org>
Subject: Re: [Speechsc] draft-ietf-speechsc-mrcpv2-00: multiple media
 sessions
Message-ID: <306066203.1065604522@[10.32.254.182]>
In-Reply-To: <BBF29C9B95E52E4DB5C29A0ACC94E83B3EEB1D@ac-exch1.eu.scansoft.com>
References:  <BBF29C9B95E52E4DB5C29A0ACC94E83B3EEB1D@ac-exch1.eu.scansoft.com
 >
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Hmmm, I think KLaus has a good point here. There may be very good reasons=20
to send separate RTP streams to different media resources for the same=20
logical session. Given that we do not want to create extra SIP dialogs and=20
have multiple MRCP sessions as a side effect of that I think that some=20
explicit matching of RTP streams and MRCP control sessions is needed.

I think fid probably fills the bill here. I'd prefer to not introduce=20
convention hacks like "i=3D" to do the matching.

Dave.

--On Wednesday, October 08, 2003 8:54 AM -0400 "Reifenrath, Klaus"=20
<Klaus.Reifenrath@Scansoft.com> wrote:

> According to Chapter 3.8 of draft-ieft-speechsc-reqts-04 MRCPv2 MUST
> allow a 1:N mapping between session and RTP channels. But the spec
> (draft-ietf-speechsc-mrcpv2-00 chapter 2 last paragraph and chapter 3.3)
> says that the media pipes MUST be shared by all media processing =
resources
> under that SIP session. Why this restriction?
>
> How would a sample for the 1:N use case look like? Especially how does =
the
> client know which RTP stream is used for ASR and which for SV/SI (if it
> does not have this info it has to send all audio data of every utterance
> to ASR and SV/SI, even if only ASR or SV/SI is activated at the same
> time)? The "i=3D" field (of the SDP session description) could be used to
> label the different media streams.
>
> Is the following sequence of messages MRCPv2 compliant?
> C->S:
>            INVITE sip:mresources@mediaserver.com SIP/2.0
>            Max-Forwards: 70
>            To: MediaServer <sip:mresources@mediaserver.com>
>            From: sarvi <sip:sarvi@cisco.com>;tag=3D1928301774
>            Call-ID: a84b4c76e66710
>            CSeq: 314163 INVITE
>            Contact: <sip: sarvi@cisco.com>
>            Content-Type: application/sdp
>            Content-Length: 142
>
>            v=3D0
>            o=3Dsarvi 2890844526 2890842809 IN IP4 126.16.64.4
>            s=3DSDP Seminar
>            i=3DA session for processing media
>            c=3DIN IP4 224.2.17.12/127
>            m=3Dcontrol 0 mrcp 01
>            m=3Dcontrol 0 mrcp 02
>            m=3Dcontrol 0 mrcp 04
>            m=3Daudio 49170 RTP/AVP 0 96
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dfmtp:96 0-15
>            a=3Dsendrecv
>
>  S->C:
>            SIP/2.0 200 OK
>            To: MediaServer <sip:mresources@mediaserver.com>
>            From: sarvi <sip:sarvi@cisco.com>;tag=3D1928301774
>            Call-ID: a84b4c76e66710
>            CSeq: 314163 INVITE
>            Contact: <sip: sarvi@cisco.com>
>            Content-Type: application/sdp
>            Content-Length: 131
>
>            v=3D0
>            o=3Dsarvi 2890844526 2890842809 IN IP4 126.16.64.4
>            s=3DSDP Seminar
>            i=3DA session for processing media
>            c=3DIN IP4 224.2.17.12/127
>            m=3Dcontrol 32416 mrcp 32AECB23433801
>            m=3Dcontrol 32416 mrcp 32AECB23433802
>            m=3Dcontrol 32416 mrcp 32AECB23433804
>            m=3Daudio 48260 RTP/AVP 0 96
>            i=3DASR
>            c=3DIN IP4 224.2.17.13/127
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dfmtp:96 0-15
>            a=3Drecvonly
>            m=3Daudio 48260 RTP/AVP 0 96
>            i=3DTTS
>            c=3DIN IP4 224.2.17.14/127
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dsendonly
>            m=3Daudio 48260 RTP/AVP 0 96
>            i=3DSV
>            c=3DIN IP4 224.2.17.15/127
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dfmtp:96 0-15
>            a=3Drecvonly
>
>     C->S:
>            ACK sip:mrcp@mediaserver.com SIP/2.0
>            Max-Forwards: 70
>            To: MediaServer <sip:mrcp@mediaserver.com>;tag=3Da6c85cf
>            From: Sarvi <sip:sarvi@cisco.com>;tag=3D1928301774
>            Call-ID: a84b4c76e66710
>            CSeq: 314164 ACK
>            Content-Length: 0
>
> And what about the following message sequence?
>
> C->S:
>            INVITE sip:mresources@mediaserver.com SIP/2.0
>            Max-Forwards: 70
>            To: MediaServer <sip:mresources@mediaserver.com>
>            From: sarvi <sip:sarvi@cisco.com>;tag=3D1928301774
>            Call-ID: a84b4c76e66710
>            CSeq: 314163 INVITE
>            Contact: <sip: sarvi@cisco.com>
>            Content-Type: application/sdp
>            Content-Length: 142
>
>            v=3D0
>            o=3Dsarvi 2890844526 2890842809 IN IP4 126.16.64.4
>            s=3DSDP Seminar
>            i=3DA session for processing media
>            c=3DIN IP4 224.2.17.12/127
>            m=3Dcontrol 0 mrcp 01
>            m=3Dcontrol 0 mrcp 02
>            m=3Dcontrol 0 mrcp 04
>            m=3Daudio 49170 RTP/AVP 0 96
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dfmtp:96 0-15
>            a=3Dsendrecv
>
>  S->C:
>            SIP/2.0 200 OK
>            To: MediaServer <sip:mresources@mediaserver.com>
>            From: sarvi <sip:sarvi@cisco.com>;tag=3D1928301774
>            Call-ID: a84b4c76e66710
>            CSeq: 314163 INVITE
>            Contact: <sip: sarvi@cisco.com>
>            Content-Type: application/sdp
>            Content-Length: 131
>
>            v=3D0
>            o=3Dsarvi 2890844526 2890842809 IN IP4 126.16.64.4
>            s=3DSDP Seminar
>            i=3DA session for processing media
>            m=3Dcontrol 32416 mrcp 32AECB23433801
>            c=3DIN IP4 224.2.17.13/127
>            m=3Dcontrol 32416 mrcp 32AECB23433802
>            c=3DIN IP4 224.2.17.14/127
>            m=3Dcontrol 32416 mrcp 32AECB23433804
>            c=3DIN IP4 224.2.17.15/127
>            m=3Daudio 48260 RTP/AVP 0 96
>            i=3DASR
>            c=3DIN IP4 224.2.17.13/127
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dfmtp:96 0-15
>            a=3Drecvonly
>            m=3Daudio 48260 RTP/AVP 0 96
>            i=3DTTS
>            c=3DIN IP4 224.2.17.14/127
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dsendonly
>            m=3Daudio 48260 RTP/AVP 0 96
>            i=3DSV
>            c=3DIN IP4 224.2.17.15/127
>            a=3Drtpmap:0 pcmu/8000
>            a=3Drtpmap:96 telephone-event/8000
>            a=3Dfmtp:96 0-15
>            a=3Drecvonly
>
>     C->S:
>            ACK sip:mrcp@mediaserver.com SIP/2.0
>            Max-Forwards: 70
>            To: MediaServer <sip:mrcp@mediaserver.com>;tag=3Da6c85cf
>            From: Sarvi <sip:sarvi@cisco.com>;tag=3D1928301774
>            Call-ID: a84b4c76e66710
>            CSeq: 314164 ACK
>            Content-Length: 0
>
> _______________________________________________
> Speechsc mailing list
> Speechsc@ietf.org
> https://www1.ietf.org/mailman/listinfo/speechsc



------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
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Date: Thu, 16 Oct 2003 17:09:44 -0400
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Thread-Topic: Want to be a document editor?
Thread-Index: AcOUKnT6vdskc6ffRLqi3BpWrtMU9g==
From: "Eric Burger" <eburger@snowshore.com>
To: "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
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    Sunday, November 10, 2003
    =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=

    1330-1500  Editor's Training -- Conrad C

    Training for current or aspiring IETF document
    editors.  Covers the roles and responsibilities=20
    of a document editor, and includes advice on=20
    producing a high-quality IETF specification.


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Subject: [Speechsc] MRCPv2  recognition-complete question
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Hi,

I've got a question on the results returned in a RECOGNITION-COMPLETE 
message.  The example in section 8.8 shows these results being returned:

           <?xml version="1.0"?> 
           <result x-model="http://IdentityModel" 
             xmlns:xf="http://www.w3.org/2000/xforms" 
             grammar="session:request1@form-level.store> 
                <interpretation> 
                     <xf:instance name="Person"> 
                       <Person> 
                           <Name> Andre Roy </Name> 
                       </Person> 
                     </xf:instance> 
                     <input>   may I speak to Andre Roy </input> 
                </interpretation> 
           </result> 

Where does the x-model come from?  Nothing in any of the grammars says 
that this grammar (request1@form-level.store) should be populating an 
xforms model named "IdentityModel", nor that part of the results for the 
rule named "request" should be used to populate the instance's <Person>. 
How is this expected to happen?

Thanks,
Jeff
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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">I've got a question on the results returned
in a RECOGNITION-COMPLETE message. &nbsp;The example in section 8.8 shows
these results being returned:</font>
<br>
<br><font size=3><tt>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;?xml
version=&quot;1.0&quot;?&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &lt;result x-model=&quot;http://IdentityModel&quot;
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; xmlns:xf=&quot;http://www.w3.org/2000/xforms&quot;
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; grammar=&quot;session:request1@form-level.store&gt;
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;interpretation&gt;
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&lt;xf:instance name=&quot;Person&quot;&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &lt;Person&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &lt;Name&gt; Andre Roy &lt;/Name&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &lt;/Person&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&lt;/xf:instance&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&lt;input&gt; &nbsp; may I speak to Andre Roy &lt;/input&gt; <br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;/interpretation&gt;
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &lt;/result&gt; <br>
</tt></font>
<br><font size=2 face="sans-serif">Where does the x-model come from? &nbsp;Nothing
in any of the grammars says that this grammar (request1@form-level.store)
should be populating an xforms model named &quot;IdentityModel&quot;, nor
that part of the results for the rule named &quot;request&quot; should
be used to populate the instance's &lt;Person&gt;. &nbsp;How is this expected
to happen?</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br><font size=2 face="sans-serif">Jeff</font>
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Anything other than:

Statii on various documents at IESG
MRCPv2 Draft

???


Respond to me or Dave PRIVATELY, please.

--
- Eric


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Date: Sun, 26 Oct 2003 08:06:42 -0500
From: "David R. Oran" <oran@cisco.com>
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Subject: [Speechsc] FYI: W3C working on tmeed text
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Marginally relevant to speechsc, forwarded just for people's information.

Dave.

------------ Forwarded Message ------------
Date: Friday, October 24, 2003 4:46 PM -0700
From: Dave Singer <singer@apple.com>
To: gen-sys@advent.ee.columbia.edu, avt@ietf.org, 
3GPP_TSG_SA_WG4@LIST.ETSI.ORG
Cc: "Glenn A. Adams" <glenn@xfsi.com>
Subject: [AVT] Soliciting Feedback on Timed Text (TT) Authoring Format 1.0 
Use Cases  and Requirements

Friends in MPEG-4 systems, IETF AVT, and 3GPP SA4.

The timed text working group at W3C is working, as you know, on an
authoring format for Timed Text.  There is a public document on which your
comments are solicited, as we reach this mile-stone in the work. I forward
the formal request from the chairman, below.

* * * * * * * * * * * * * *


The Timed Text Working Group invites your working group(s) to comment on
the 2nd public draft of "Timed Text (TT) Authoring Format 1.0 Use Cases and
Requirements" [1]. We expect to accept and process comments that we receive
on or before Dec 1, after which time the Timed Text Working Group intends
to publish this document (possibly with changes to address comments) as a
"Working Group Note".

Please send comments to the public reflector <public-tt@w3.org>, which is
archived at [2].

Regards,
Glenn Adams, Chair TT WG

[1] http://www.w3.org/TR/2003/WD-tt-af-1-0-req-20030915/
[2] http://lists.w3.org/Archives/Public/public-tt/
-- 
David Singer
Apple Computer/QuickTime

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