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To: "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
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Subject: [Speechsc] Active-Request-Id-List
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Section 5.4.2 of the spec (active-request-id-list) says that when no id is 
sent, it's assumed to be a wildcard and the method applies to all pending 
and in-progress requests, which is reasonable.  But the example says that 
when a wildcard stop is sent to a synthesizer, "no SPEAK-COMPLETE or 
RECOGNITION-COMPLETE events will be sent..." - is the 
"RECOGNITION-COMPLETE" part a typo?  I'm not sure why a request to a 
synthesizer would affect a recognizer.

Thanks,
Jeff

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Where is the timeout for the "too-much-speech-timeout" set?  It's one of 
the completion causes for RECOGNITION message (008), but there don't 
appear to be any messages that can actually set this value.  It looks 
similar to the recognition-timeout; is too-much-speech-timeout redundant?

Thanks,
Jeff

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To: "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
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Subject: [Speechsc] Speed vs Accuracy and Sensitivity Threshold values
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Sections 8.4.2 and 8.4.3 describe the Sensitivity-Level  and 
Speed-vs-Accuracy settings, but they don't specify the ranges nor the 
default value.  To be consistent with other parameters, might I suggest 
the range be between 0 and 100 (which maps nicely to VoiceXML 2.0's range 
of 0.0 to 1.0), and default values that are either platform-specific, as 
is the case with confidence threshold, or just 50.

Thanks,
Jeff

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>From: Jeff Kusnitz <jk@us.ibm.com>
>Date: Tue, 13 Jan 2004 15:50:05 -0800
>
>Sections 8.4.2 and 8.4.3 describe the Sensitivity-Level  and
Speed-vs-Accuracy settings, but they don't specify the ranges nor the
default value.  To be consistent with=20
>other parameters, might I suggest the range be between 0 and 100 (which
maps nicely to VoiceXML 2.0's range of 0.0 to 1.0), and default values
that are either platform-
>specific, as is the case with confidence threshold, or just 50.

If mapping nicely to VoiceXML ranges is desired (it should be), then why
not just use 0.0 to 1.0? =20
Separately, the same can be said for confidence thresholds (they should
be 0.0 to 1.0 - if a native engine uses 0 to 100, it's an easy
translation).

Cheers,
Patrick Bennett
Interactive Intelligence, Inc.


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As currently defined in both the v1 and the v2 specifications, is allowed 
in only SET-PARAMS/GET-PARAMS requests and responses.  Is this 
intentional?  I can think of instances where I'd like to use this header 
on other requests (SPEAK, RECOGNIZE, DEFINE-GRAMMAR, etc).  I realize I 
can work around this limitation by calling SET-PARAMS before my 
SPEAK/RECOGNIZE/etc request with the appropriate vendor-specific-parameter 
value, and then calling GET-PARAMS afterwards, but that requires three 
request/responses, rather than one.

Thanks,
Jeff

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--On Thursday, January 15, 2004 2:02 PM -0800 Jeff Kusnitz <jk@us.ibm.com> 
wrote:

> As currently defined in both the v1 and the v2 specifications, is allowed
> in only SET-PARAMS/GET-PARAMS requests and responses.  Is this
> intentional?  I can think of instances where I'd like to use this header
> on other requests (SPEAK, RECOGNIZE, DEFINE-GRAMMAR, etc).  I realize I
> can work around this limitation by calling SET-PARAMS before my
> SPEAK/RECOGNIZE/etc request with the appropriate
> vendor-specific-parameter  value, and then calling GET-PARAMS afterwards,
> but that requires three  request/responses, rather than one.
>
the tradeoff seems to be the extra round trip versus the need to deal with 
extra failure modes of all the other commands on which this could be 
piggy-backed. One optimization that makes sense to me is to piggyback the 
param values that would be returned on GET-PARAMS onto the response to the 
SET-PARAMS request.

If the parameters are "sticky" (not clear from teh spec that they are but I 
would assume so) then the extra round trip isn't terribly painful and it 
avoids the extra complexity of figuring out how to handle error cases on 
each request.

I don't feel strongly about this, though. Sarvi and others, what do you 
think?

Dave.
> Thanks,
> Jeff
>
> _______________________________________________
> Speechsc mailing list
> Speechsc@ietf.org
> https://www1.ietf.org/mailman/listinfo/speechsc





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From: Cullen Jennings <fluffy@cisco.com>
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What's the status of NLSML since MRCP normatively depends on it? Is there a
later version than what is at http://www.w3.org/TR/nl-spec/  ?

It looks almost like abandoned work - the draft looks like a pretty rough
early draft and was last updated in 2000. Is work proceeding on this? Is
there a more recent version I should be reading?

Thanks,  Cullen




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From: Cullen Jennings <fluffy@cisco.com>
To: SPEECHSC <speechsc@ietf.org>, Eric Burger <eburger@snowshore.com>,
        Dave Oran <oran@cisco.com>
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I'm trying to figure out the scope of what I should or should not use MRCPv2
for so that I can try and decide what might be missing from it.

Consider the use case of an IVR build with a separated media device and
controller. The media device plays audio prompts, reports DTMF input up to
the controller, can stop audio playback when there is DTMF input, and can
record audio it receives to files for later playback. It does not do any
TTS, ASR, or speaker verification.

Is this a use case we want to solve with MRCPv2 or should a different
protocol be used for this?

Thanks, Cullen


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



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Date: Sun, 18 Jan 2004 23:01:44 -0800
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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The requirements you have mentioned have been brought up individually by 
different people.
The DTMF recognition, Prompt playout, and the barge-in scenario are 
should be possible with MRCPv2.
The recording capability is currently not in the draft, but has been 
brought up in discussions is being worked on. This is now currently 
available in a MRCPv1 extension draft by Dan Burnett from Nuance.

Sarvi

Cullen Jennings wrote:

>I'm trying to figure out the scope of what I should or should not use MRCPv2
>for so that I can try and decide what might be missing from it.
>
>Consider the use case of an IVR build with a separated media device and
>controller. The media device plays audio prompts, reports DTMF input up to
>the controller, can stop audio playback when there is DTMF input, and can
>record audio it receives to files for later playback. It does not do any
>TTS, ASR, or speaker verification.
>
>Is this a use case we want to solve with MRCPv2 or should a different
>protocol be used for this?
>
>Thanks, Cullen
>
>
>_______________________________________________
>Speechsc mailing list
>Speechsc@ietf.org
>https://www1.ietf.org/mailman/listinfo/speechsc
>
>  
>


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



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Going forward, EMMA is W3C recommendation to replace NLSML. But for now 
NLSML is expected to be supported for backward compatibility.

Sarvi

Cullen Jennings wrote:

>What's the status of NLSML since MRCP normatively depends on it? Is there a
>later version than what is at http://www.w3.org/TR/nl-spec/  ?
>
>It looks almost like abandoned work - the draft looks like a pretty rough
>early draft and was last updated in 2000. Is work proceeding on this? Is
>there a more recent version I should be reading?
>
>Thanks,  Cullen
>
>
>
>
>_______________________________________________
>Speechsc mailing list
>Speechsc@ietf.org
>https://www1.ietf.org/mailman/listinfo/speechsc
>
>  
>


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



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Date: Sun, 18 Jan 2004 18:58:11 -0500
From: David Oran <oran@cisco.com>
To: Jeff Kusnitz <jk@us.ibm.com>, "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
Subject: Re: [Speechsc] Speed vs Accuracy and Sensitivity Threshold values
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--On Tuesday, January 13, 2004 3:50 PM -0800 Jeff Kusnitz <jk@us.ibm.com> 
wrote:

> Sections 8.4.2 and 8.4.3 describe the Sensitivity-Level  and
> Speed-vs-Accuracy settings, but they don't specify the ranges nor the
> default value.  To be consistent with other parameters, might I suggest
> the range be between 0 and 100 (which maps nicely to VoiceXML 2.0's range
> of 0.0 to 1.0), and default values that are either platform-specific, as
> is the case with confidence threshold, or just 50.
>
> Thanks,
> Jeff
>
Seems reasonable? any objections?
Dave.


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https://www1.ietf.org/mailman/listinfo/speechsc



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From: Patrick Bennett <patrick.bennett@inin.com>
To: David Oran <oran@cisco.com>
Date: Mon, 19 Jan 2004 12:00:15 -0500
Subject: Re: [Speechsc] Speed vs Accuracy and Sensitivity Threshold values
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David Oran wrote:

> --On Tuesday, January 13, 2004 3:50 PM -0800 Jeff Kusnitz 
> <jk@us.ibm.com> wrote:
>
>> Sections 8.4.2 and 8.4.3 describe the Sensitivity-Level  and
>> Speed-vs-Accuracy settings, but they don't specify the ranges nor the
>> default value.  To be consistent with other parameters, might I suggest
>> the range be between 0 and 100 (which maps nicely to VoiceXML 2.0's 
>> range
>> of 0.0 to 1.0), and default values that are either platform-specific, as
>> is the case with confidence threshold, or just 50.
>>
>> Thanks,
>> Jeff
>>
> Seems reasonable? any objections?
> Dave.

This is a draft specification after all, so why not just make some of 
those other parameters consistent with VoiceXML 2.0 and require 0.0 to 
1.0 for all of them (Confidence-Threshold on particular)?
Is there really much benefit for a MRCP 'user' in leaving these ranges 
undefined and vendor-specific?

Patrick Bennett



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Subject: Re: [Speechsc] Speed vs Accuracy and Sensitivity Threshold values
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I am fine wiht either 0.0 to 1.0 or 0 to 100. My leaning is toward 
keeping the set of header fields that have a parallel in VXML to have 
the same range and default.
The default value is already specified as platform specific, and I would 
prefer to leave it as is, since different reco vendors have different 
default values or for that matter different charactersitics for the 
range of values.

Sarvi

David Oran wrote:

> --On Tuesday, January 13, 2004 3:50 PM -0800 Jeff Kusnitz 
> <jk@us.ibm.com> wrote:
>
>> Sections 8.4.2 and 8.4.3 describe the Sensitivity-Level  and
>> Speed-vs-Accuracy settings, but they don't specify the ranges nor the
>> default value.  To be consistent with other parameters, might I suggest
>> the range be between 0 and 100 (which maps nicely to VoiceXML 2.0's 
>> range
>> of 0.0 to 1.0), and default values that are either platform-specific, as
>> is the case with confidence threshold, or just 50.
>>
>> Thanks,
>> Jeff
>>
> Seems reasonable? any objections?
> Dave.
>
>
> _______________________________________________
> Speechsc mailing list
> Speechsc@ietf.org
> https://www1.ietf.org/mailman/listinfo/speechsc
>


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



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The problem with using the same values as VoiceXML uses is that VoiceXML 
values are 0.0 to 1.0, while MRCP are defined to be integer (the 
production for them is "1*DIGIT CRLF").  Confidence-level is specified in 
MRCP already as 0-100, while the VoiceXML counterpart is 0.0-1.0.

Thanks,
Jeff


> I am fine with either 0.0 to 1.0 or 0 to 100. My leaning is toward 
> keeping the set of header fields that have a parallel in VXML to have 
> the same range and default.
> The default value is already specified as platform specific, and I would 

> prefer to leave it as is, since different reco vendors have different 
> default values or for that matter different charactersitics for the 
> range of values.


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



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From: Patrick Bennett <patrick.bennett@inin.com>
To: Sarvi Shanmugham <sarvi@cisco.com>
Date: Mon, 19 Jan 2004 14:29:08 -0500
Subject: Re: [Speechsc] Speed vs Accuracy and Sensitivity Threshold values
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I'm pretty late in the MRCP process so I'm sorry if this comes across as 
rude, but, what is the point of a vendor-neutral protocol like MRCP if 
an MRCP-client has to always be aware of the vendor which it is 
communicating with?  Wouldn't it make more sense to define a required 
value range and have the vendors provide a reasonable mapping?  
Considering Nuance typically uses 0 to 100 and SpeechWorks/VXML uses 0.0 
to 1.0, why not just use 0.0 to 1.0 and have Nuance (or other vendors) 
do the mapping in their MRCP servers?  The 0 to 1 range (good or not) 
definitely seems to be a bit more 'standard' (if there can be such a 
thing).  Providing so many opening for vendor specific interpretations 
severely (at least IMO) undermines the usefulness of MRCP.

Sarvi Shanmugham wrote:

> I am fine wiht either 0.0 to 1.0 or 0 to 100. My leaning is toward 
> keeping the set of header fields that have a parallel in VXML to have 
> the same range and default.
> The default value is already specified as platform specific, and I 
> would prefer to leave it as is, since different reco vendors have 
> different default values or for that matter different charactersitics 
> for the range of values.
>
> Sarvi
>
> David Oran wrote:
>
>> --On Tuesday, January 13, 2004 3:50 PM -0800 Jeff Kusnitz 
>> <jk@us.ibm.com> wrote:
>>
>>> Sections 8.4.2 and 8.4.3 describe the Sensitivity-Level  and
>>> Speed-vs-Accuracy settings, but they don't specify the ranges nor the
>>> default value.  To be consistent with other parameters, might I suggest
>>> the range be between 0 and 100 (which maps nicely to VoiceXML 2.0's 
>>> range
>>> of 0.0 to 1.0), and default values that are either 
>>> platform-specific, as
>>> is the case with confidence threshold, or just 50.
>>>
>>> Thanks,
>>> Jeff
>>>
>> Seems reasonable? any objections?
>> Dave.
>>


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From: "Wyss, Felix" <FelixW@inin.com>
To: "Jeff Kusnitz" <jk@us.ibm.com>, "Sarvi Shanmugham" <sarvi@cisco.com>
Cc: "David Oran" <oran@cisco.com>,
        "IETF SPEECHSC \(E-mail\)" <speechsc@ietf.org>,
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Date: Mon, 19 Jan 2004 14:47:09 -0500
Subject: RE: [Speechsc] Speed vs Accuracy and Sensitivity Threshold values
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A range of 0...100 seems very arbitrary and limits the possible
precision (not that this matters in this case, but nevertheless).  Why
not 0...1000 or 0...1000000?  In my opinion, either the range should be
0.0 to 1.0 with arbitrary precision or the value should always be sent
with the upper bound (i.e. as fraction), for example: "12/100",
"437/1000"

--Felix

> -----Original Message-----
> From: speechsc-admin@ietf.org=20
> [mailto:speechsc-admin@ietf.org] On Behalf Of Jeff Kusnitz
> Sent: Monday, January 19, 2004 14:23
> To: Sarvi Shanmugham
> Cc: David Oran; IETF SPEECHSC (E-mail); speechsc-admin@ietf.org
> Subject: Re: [Speechsc] Speed vs Accuracy and Sensitivity=20
> Threshold values
>=20
> The problem with using the same values as VoiceXML uses is=20
> that VoiceXML values are 0.0 to 1.0, while MRCP are defined=20
> to be integer (the production for them is "1*DIGIT CRLF"). =20
> Confidence-level is specified in MRCP already as 0-100, while=20
> the VoiceXML counterpart is 0.0-1.0.
>=20
> Thanks,
> Jeff
>=20
>=20
> > I am fine with either 0.0 to 1.0 or 0 to 100. My leaning is toward=20
> > keeping the set of header fields that have a parallel in=20
> VXML to have=20
> > the same range and default.
> > The default value is already specified as platform specific, and I=20
> > would
>=20
> > prefer to leave it as is, since different reco vendors have=20
> different=20
> > default values or for that matter different charactersitics for the=20
> > range of values.
>=20
>=20
> _______________________________________________
> Speechsc mailing list
> Speechsc@ietf.org
> https://www1.ietf.org/mailman/listinfo/speechsc
>=20
>=20


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You bring up an interesting point on the stickiness..  In a couple of 
places, a distinction is made between values sent on a SET-PARAMS request 
vs other.  For example, in cache-control (5.4.11), it says this:

        If the directives are sent on a SET-PARAMS method, it 
        SHOULD apply for all requests for documents the media server may 
        make in that session. If the directives are sent on any other 
        messages they MUST only apply to document requests the media
        server needs to make for that method.

And similarly, in voice-parameters (7.4.5):

        These header fields MAY BE sent in SET-PARAMS/GET-PARAMS request
        to define/get default values for the entire session or MAY BE
        sent in the SPEAK request to define default values for that
        speak request.  Further more these attributes can be part of the
        speech text marked up in SML. 

I had assumed (oopsie?) that this would be applicable in all places - a 
value set via SET-PARAMS is retained for the duration of a session, while 
a value set on some other request was valid only for that request.  It 
seemed fairly natural as well, but perhaps it should be spelled out in the 
SET-PARAMS sections?

Jeff

> > As currently defined in both the v1 and the v2 specifications, is 
allowed
> > in only SET-PARAMS/GET-PARAMS requests and responses.  Is this
> > intentional?  I can think of instances where I'd like to use this 
header
> > on other requests (SPEAK, RECOGNIZE, DEFINE-GRAMMAR, etc).  I realize 
I
> > can work around this limitation by calling SET-PARAMS before my
> > SPEAK/RECOGNIZE/etc request with the appropriate
> > vendor-specific-parameter  value, and then calling GET-PARAMS 
afterwards,
> > but that requires three  request/responses, rather than one.
> >
> the tradeoff seems to be the extra round trip versus the need to deal 
with 
> extra failure modes of all the other commands on which this could be 
> piggy-backed. One optimization that makes sense to me is to piggyback 
the 
> param values that would be returned on GET-PARAMS onto the response to 
the 
> SET-PARAMS request.
> 
> If the parameters are "sticky" (not clear from teh spec that they are 
but I 
> would assume so) then the extra round trip isn't terribly painful and it 

> avoids the extra complexity of figuring out how to handle error cases on 

> each request.
> 
> I don't feel strongly about this, though. Sarvi and others, what do you 
> think?



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Date: Tue, 20 Jan 2004 10:05:23 -0500
Subject: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-01.txt
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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		: Media Resource Control Protocol Version 2(MRCPv2)
	Author(s)	: S. Shanmugham
	Filename	: draft-ietf-speechsc-mrcpv2-01.txt
	Pages		: 88
	Date		: 2004-1-19
	
This document describes a proposal for a Media Resource Control 
Protocol Version 2(MRCPv2) and aims to meet the requirements 
specified in the SPEECHSC working group requirements document. It is 
based on the Media Resource Control Protocol (MRCP), also called 
MRCPv1 developed jointly by Cisco Systems, Inc., Nuance 
Communications, and Speechworks Inc.  
The MRCPv2 protocol will control media service resources like speech 
synthesizers, recognizers, signal generators, signal detectors, fax 
servers etc. over a network. This protocol depends on a session 
management protocol such as the Session Initiation Protocol (SIP) to 
establish a separate MRCPv2 control session between the client and 
the media server. It also depends on SIP to establish the media pipe 
and associated parameters between the media source or sink and the 
media server. Once this is done, the MRCPv2 protocol exchange can 
happen over the control session established above allowing the 
client to command and control the media processing resources that 
may exist on the media server.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-01.txt

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	"multipart" MIME messages (i.e. documents which have been split
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implementation to automatically retrieve the ASCII version of the
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--NextPart
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ENCODING mime
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	directory="internet-drafts"

Content-Type: text/plain
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--OtherAccess--

--NextPart--



_______________________________________________
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https://www1.ietf.org/mailman/listinfo/speechsc



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Subject: RE: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-01.txt
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Tout va tr=E8s bien Madame La Marquise ... tralalala ... lala ... lal=E8r=
e


-----Message d'origine-----
De : speechsc-admin@ietf.org [mailto:speechsc-admin@ietf.org]De la part
de Internet-Drafts@ietf.org
Envoy=E9 : mardi 20 janvier 2004 16:05
=C0 : IETF-Announce:
Cc : speechsc@ietf.org
Objet : [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-01.txt


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		: Media Resource Control Protocol Version 2(MRCPv2)
	Author(s)	: S. Shanmugham
	Filename	: draft-ietf-speechsc-mrcpv2-01.txt
	Pages		: 88
	Date		: 2004-1-19

This document describes a proposal for a Media Resource Control
Protocol Version 2(MRCPv2) and aims to meet the requirements
specified in the SPEECHSC working group requirements document. It is
based on the Media Resource Control Protocol (MRCP), also called
MRCPv1 developed jointly by Cisco Systems, Inc., Nuance
Communications, and Speechworks Inc.
The MRCPv2 protocol will control media service resources like speech
synthesizers, recognizers, signal generators, signal detectors, fax
servers etc. over a network. This protocol depends on a session
management protocol such as the Session Initiation Protocol (SIP) to
establish a separate MRCPv2 control session between the client and
the media server. It also depends on SIP to establish the media pipe
and associated parameters between the media source or sink and the
media server. Once this is done, the MRCPv2 protocol exchange can
happen over the control session established above allowing the
client to command and control the media processing resources that
may exist on the media server.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-01.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 messag=
e.

Internet-Drafts are also available by anonymous FTP. Login with the usern=
ame
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-speechsc-mrcpv2-01.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-mrcpv2-01.txt".

NOTE:	The mail server at ietf.org can return the document in
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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.


Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.


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



From exim@www1.ietf.org  Wed Jan 21 09:34:56 2004
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To: <speechsc@ietf.org>
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From: "Lawrence Reisler" <Larry_Reisler@nmss.com>
Date: Wed, 21 Jan 2004 16:09:31 +0200
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 13, 2003) at 01/21/2004 09:10:33 AM
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Subject: [Speechsc] VoiceXML fetchaudiominimum
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Given the current draft spec (draft-ietf-speechsc-mrcpv2-01), I was
wondering how the VoiceXML fetcaudiominimum property could be implemented.
This requires a prompt, once it has begun, to play a minimum number of
seconds before completing.

You could in theory, use the CONTROL method to modify the Speak-Length
parameter when the fetch completes, but according to the spec the length of
time is "relative to the current speaking point in the currently active
SPEAK request" as opposed to an absolute length from the beginning of the
prompt.

Thanks,
Larry



_______________________________________________
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Subject: Re: [Speechsc] VoiceXML fetchaudiominimum
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From: Jeff Kusnitz <jk@us.ibm.com>
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I confess I've not read the new draft, but I don't see why this isn't 
something the client would handle on it's own - just start playing the 
fetchaudio, making a note of the time, and when the fetch completes, if 
fetchaudiominimum has not yet elapsed wait until it has and then issue a 
STOP request.

Or make it simpler and don't even bother with the TTS resource.  Playing 
an audio file is something the client (or a generic audio playing 
resource) could handle on it's own..

Jeff

"Lawrence Reisler" <Larry_Reisler@nmss.com> wrote on 01/21/2004 06:09:31 
AM:

> 
> Given the current draft spec (draft-ietf-speechsc-mrcpv2-01), I was
> wondering how the VoiceXML fetcaudiominimum property could be 
implemented.
> This requires a prompt, once it has begun, to play a minimum number of
> seconds before completing.
> 
> You could in theory, use the CONTROL method to modify the Speak-Length
> parameter when the fetch completes, but according to the spec the length 
of
> time is "relative to the current speaking point in the currently active
> SPEAK request" as opposed to an absolute length from the beginning of 
the
> prompt.
> 
> Thanks,
> Larry


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Date: Thu, 22 Jan 2004 07:10:30 -0800
From: Cullen Jennings <fluffy@cisco.com>
To: SPEECHSC <speechsc@ietf.org>, Sarvi Shanmugham <sarvi@cisco.com>
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Subject: [Speechsc] mrcpv2-01 comments
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First, let me say I like it, thanks for all the hard work in. A bunch of
these comments may be that I just don't understand it well enough but here
are some thoughts/questions ....

When a devices fetches some document, when does it get rid of it.

Confusing how it is decided what parameters get set in SSML and what in MRCP
and what can be done in both. I don't really like the both solutions.

No audioplayer definition.

Section 2.2
Doe the user need to be mrcpv2 or is this just an example

Section 3.1, example 1, why bother opening a session before getting any
resources on it. 

Section 3.2 - defines  resource type of audioplayer but this is not defined
in the document 

Why did you pick a port of 9 when the port was unknown. Seems like 0 or
2^16-1 would be better?

If the server is on the inside of a NAT and the client on the outside, can
all of this work? Seems like you want the MRCPv2 control channel to open the
other direction. 

Example 3
I'm a little confused on how the listen ports on the TCP stuff work -
particularly when there are multiple of them. If I get two m control lines
that both have 32416 as the listen port at the far side, does that mean I
form two TCP connection to a TCP server listening on that port or I always
use one if it is the same port. If an app was talking to multiple things,
would it need to listen on a bunch of different TCP ports. It might be fine
how this all works but I think a little more text is needed.

Section 3.3 

Why invent cmid, couldn't you just reuse mid.


Section 5

Why do you need to discuss character sets such as ISO 8859-1. Seems like
saying UTF-8 would be enough.

<big issue> It looks like the grammar is set up such there is no easy way to
tell if a start-line is a request-line, event-line, or response-line when
parsing. I think this needs to be fixed either by changing the ABNF or
changing the syntax.

Section 5.4
I think line folding is an awful idea, slows down parsers, adds many bugs,
and has no known useful purpose. The fact that other IETF protocols do is
not a known useful purpose.

Allowing both a Header: a,b,c syntax and a multiple Headers of same type was
a bad idea in SIP (see above comment) - I would pick one or the other.

Content-Base: Do you really need to support relative URL at this level? I
understand some specific XML body having a base but can't you just use
absolute URLs at the MRCPv2 level. It's hard to get all the relative stuff
correct particularly when you have ACL based on URL.

Content-Encoding: Add that binary MUST be supported and base64 SHOULD NOT be
used. This is a 8bit safe transport.

Logging-tag - I can't decide if I love this or think it is goofy.

Section 7 - point 2 says basicsynth but I think you called it simplesynth in
section 3.2 

Kill-On-Barge-In - Do you also need some sort of pause-on-barge

When it is a numeric-speech-unit, what is the unit? If undefined, add text
to explain.

page 31 -  Prosody Parameters - do we really need this here if it is also in
SSML.

Page 51 
Waveform-URL: What format. There are tons of various formats I can put in
.wav file - I would rather not use file extensions to figure out formats.

DTMF-Term-Char - I don't get this. This is something like # after typing in
a PIN. This and the DTMF inter digit stuff seem like they should be in the
reco grammar not the MRCP.

Page 57
Can you use a cid URL instead of making up the session URL.

Section 8.8 - if a grammar is compiled, how long is it kept

Section 11 - Is this normative or not. If it conflicts with text which one
is definitive.



Some minor NITS

Lots of formatting issues.

Abstract probably does not need companies that developed MRCPv1.

All the example domain names need to be moved to example.com or such

Section 4 - after defining HX.Y notation, I don't think you use it

Section 6 - bunch of Word got you with " formatting issues.

Section 7 - this would be better drawn as a state machine instead of as a
flow. I am glad to send ASCII art if you want.

Page 32 - top - have SML when I think you mean SSML

Page 34 - "A - value" might be better to say "A negative value"


Page 58 - example 2. Has session URL to local  content but that is not shown
in the example.

Page 59 - reference to NLSML has ref number of 11 but in the bibliography it
is 12.

Section 10
#4 should move to 3261

#8 to #14 have the word " problems

Page 83 - delta-seconds does not seem to be defined


Page 86 numeric-speech-length (and a bunch of others) are redefines of stuff
on page 85


Nice work, Easy to read, Thanks, Cullen


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Subject: [Speechsc] Re: mrcpv2-01 comments
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asnwers/comments inline.

Cullen Jennings wrote:

>First, let me say I like it, thanks for all the hard work in. A bunch of
>these comments may be that I just don't understand it well enough but here
>are some thoughts/questions ....
>
>When a devices fetches some document, when does it get rid of it.
>
At least for ASR grammar URI  and TTS content URI that is 
referred/fetched,  The client can specify caching parameters for the 
session and the server is supposed to honor and it decide when to get 
rid of the document from the server cache.

For content that is created/stored on the server as a result of  
commands such as RECORD, is deleted at the end of the session. Note that 
we should also have the option to specify an external URI for recording 
in which case it doesn't get deleted by the server at the end of the 
session and it left to the client to manage.

>Confusing how it is decided what parameters get set in SSML and what in MRCP
>and what can be done in both. I don't really like the both solutions.
>
Most of the SSML parameters such as prosody, etc are available in MRCP 
as headers. but the values specified in SSML take precendence. The MRCP 
header fields allows the client to specify what thedefaultvalues are if 
the SSML document doesn't specify it or if the document is not SSML(say 
plain text).


>No audioplayer definition.
>
Section 3.2 defines it.
Secition 7.0 describes this. Its essentially a synethsizer that supports 
only the <audio> tag in SSML.

>Section 2.2
>Doe the user need to be mrcpv2 or is this just an example
>
I don't understand the question. An MRCPv2 server or client is just a 
SIP UA which also has some additional functionality.

>Section 3.1, example 1, why bother opening a session before getting any
>resources on it. 
>
Not required. It was left there to just point out that we could 
"possibly" start with 0 channels and then add from there. But there have 
been too many questions on this. I will remove it from the next version, 
to avoid confusion.

>Section 3.2 - defines  resource type of audioplayer but this is not defined
>in the document 
>
Well these are synthesizer(Section 7.0) of varying levels of capability 
in their SSML support. I think I should put some forward references to 
Section 7.0 for these resources.

>Why did you pick a port of 9 when the port was unknown. Seems like 0 or
>2^16-1 would be better?
>
Apparently, thats the NULL port and was recommended for use on the 
alias. Zero was my initial choice, but said to be not the right choice.

>If the server is on the inside of a NAT and the client on the outside, can
>all of this work? Seems like you want the MRCPv2 control channel to open the
>other direction. 
>
This may be a problem for the TCP control channel. Suggestions?

We could allow as an option, for the client to specify the TCP port it 
will be listening to (anything other than port 9), in which case the 
server needs to open the TCP pipe toqwards the client . Any thoughts?

There is also another problem that has been brought to my attention, 
whose solution may be related.  3PCC of resources.

When an MRCPv2 client is setting up a pipe between a resource and 
another endpoint, It may not have the an initial m=audio line to offer, 
may be  because it is a part of a 3pcc setup in progress.
So looks like we need to consider a possibility, where the initial offer 
for an m=audio line may have to come from the server. This might require 
the MRCPv2 client to start with an INVITE to the server with no offer 
and wait for the server to provide the first offer. In this case, we 
need an alternative mechanism of telling the server, what 
resources-types  we need, since we  would not be allowed to  have 
m=control lines in the  INVITE anyway.
I propose that we allow an XML body of some sort containing list of 
resources need, so that  server can make the initial offer with the 
appropriate  m=control lines and m=audio lines in it.

>Example 3
>I'm a little confused on how the listen ports on the TCP stuff work -
>particularly when there are multiple of them. If I get two m control lines
>that both have 32416 as the listen port at the far side, does that mean I
>form two TCP connection to a TCP server listening on that port or I always
>use one if it is the same port. If an app was talking to multiple things,
>would it need to listen on a bunch of different TCP ports. It might be fine
>how this all works but I think a little more text is needed.
>
The protocol just returns the port to talk to open a TCP pipe. between a 
client and a server, if there is already one or more TCP pipes open, 
they can be shared across multiple MRCP sessions that may be active 
between the two.
section 3.2 and 3.4

>Section 3.3 
>
>Why invent cmid, couldn't you just reuse mid.
>
Each m-line can have an mid. But what we are trying to do here is tie 
the m-line with an mid to another m=control line, essentially saying 
this media resource should feed from/to that m=audio line.
I cannot use the mid attribute with the same mid in the m=control line 
that will fail the mid uniqueness criteria.
The option is the group id defined by the same spec as the mid, that 
defines 2 specific group semantics which both don't seem to apply here.
Do you have something else in mind.


>
>
>Section 5
>
>Why do you need to discuss character sets such as ISO 8859-1. Seems like
>saying UTF-8 would be enough.
>
I think ISO 8859 is needed for ASR and TTS content for oriental and 
other language support.

><big issue> It looks like the grammar is set up such there is no easy way to
>tell if a start-line is a request-line, event-line, or response-line when
>parsing. I think this needs to be fixed either by changing the ABNF or
>changing the syntax.
>
Not sure what you mean here.
The request line has a method-name and no state
The response has no method name ans a status and a state.
The event has a method name and a state  and no status.
Could you ellaborate.

>Section 5.4
>I think line folding is an awful idea, slows down parsers, adds many bugs,
>and has no known useful purpose. The fact that other IETF protocols do is
>not a known useful purpose.
>
>Allowing both a Header: a,b,c syntax and a multiple Headers of same type was
>a bad idea in SIP (see above comment) - I would pick one or the other.
>  
>
Need to get back  to you on this.

>Content-Base: Do you really need to support relative URL at this level? I
>understand some specific XML body having a base but can't you just use
>absolute URLs at the MRCPv2 level. It's hard to get all the relative stuff
>correct particularly when you have ACL based on URL.
>  
>
I think this is an optimization. In VXML documents there is a base URL,  
which needs to be applied for relative URI references that may exist 
within GRXML or SSML documents. The client in many cases does not parse 
through the GRXML or SSML documents to expand such cases before 
providing to the Server.

>Content-Encoding: Add that binary MUST be supported and base64 SHOULD NOT be
>used. This is a 8bit safe transport.
>  
>
fine with me.

>Logging-tag - I can't decide if I love this or think it is goofy.
>  
>
This is a tag that we added in MRCPv1 and also have in VXML so that logs 
generated by the application server, VXML browser and the media 
processing resources can be corellated.

>Section 7 - point 2 says basicsynth but I think you called it simplesynth in
>section 3.2 
>
Will make the correction. I think symplesynth should do.

>
>Kill-On-Barge-In - Do you also need some sort of pause-on-barge
>  
>
I haven't heard any request for this so far.  Any thoughts people.

>When it is a numeric-speech-unit, what is the unit? If undefined, add text
>to explain.
>
Dont' understand.

>
>page 31 -  Prosody Parameters - do we really need this here if it is also in
>SSML.
>
To se the default values, if the SSML does not have the prosody or other 
such attributes. Or in cases where the content may not be plain text.

>
>Page 51 
>Waveform-URL: What format. There are tons of various formats I can put in
>.wav file - I would rather not use file extensions to figure out formats.
>
This was initially added for mostly trouble shooting purposes and to 
address the VXMl shadow variable after recognition. This just saves the 
utterance or wave form that was used in the recognition.
I don't mind adding headers to specify formats and such for this recording.

But note, that there is a separate RECORD method that is being proposed 
to do regular recording and stuff that will have mechanisms for the 
client to specify storage location, file formats etc.

>
>DTMF-Term-Char - I don't get this. This is something like # after typing in
>a PIN. This and the DTMF inter digit stuff seem like they should be in the
>reco grammar not the MRCP.
>
I think this was added to support VXML with DTMF only support, and the 
VXML spec allows for such things beyond SSML support.


      6.3.3 Generic DTMF Recognizer Properties

of the VXML specification

>
>Page 57
>Can you use a cid URL instead of making up the session URL.
>
Well, I thought cid URL was only valid to refer to other parts of a 
single multi-part message.
This is a case, where we have defined grammars with prior methods during 
the session. These get stored
on the server referenced by the Content-ID that came with content.
Now can we really use the cid to refer that content. I thought this was 
not allowed. If thats not the case, I have no problem using cid.

>
>Section 8.8 - if a grammar is compiled, how long is it kept
>  
>
Wtih the methods in place, grammars defined using section 8.8 are valid 
for that session. There are requests for managing server level grammars, 
but that has not been addressed yet.

>Section 11 - Is this normative or not. If it conflicts with text which one
>is definitive.
>
Section 11 is definitive. Will try to remove the conflicts as much as 
possible and make it clear this section is definitive.

>
>
>
>Some minor NITS
>
>Lots of formatting issues.
>
>Abstract probably does not need companies that developed MRCPv1.
>
>All the example domain names need to be moved to example.com or such
>
>Section 4 - after defining HX.Y notation, I don't think you use it
>
>Section 6 - bunch of Word got you with " formatting issues.
>
>Section 7 - this would be better drawn as a state machine instead of as a
>flow. I am glad to send ASCII art if you want.
>
>Page 32 - top - have SML when I think you mean SSML
>
>Page 34 - "A - value" might be better to say "A negative value"
>
>
>Page 58 - example 2. Has session URL to local  content but that is not shown
>in the example.
>
>Page 59 - reference to NLSML has ref number of 11 but in the bibliography it
>is 12.
>
>Section 10
>#4 should move to 3261
>
>#8 to #14 have the word " problems
>
>Page 83 - delta-seconds does not seem to be defined
>
>
>Page 86 numeric-speech-length (and a bunch of others) are redefines of stuff
>on page 85
>

thanks for the comments, I will try to fix them in the next release of 
the spec.

Sarvi

>
>
>Nice work, Easy to read, Thanks, Cullen
>
>
>  
>

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asnwers/comments inline.<br>
<br>
Cullen Jennings wrote:<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">First, let me say I like it, thanks for all the hard work in. A bunch of
these comments may be that I just don't understand it well enough but here
are some thoughts/questions ....

When a devices fetches some document, when does it get rid of it.</pre>
</blockquote>
At least for ASR grammar URI&nbsp; and TTS content URI that is
referred/fetched,&nbsp; The client can specify caching parameters for the
session and the server is supposed to honor and it decide when to get
rid of the document from the server cache.<br>
<br>
For content that is created/stored on the server as a result of&nbsp;
commands such as RECORD, is deleted at the end of the session. Note
that we should also have the option to specify an external URI for
recording in which case it doesn't get deleted by the server at the end
of the session and it left to the client to manage.<br>
<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Confusing how it is decided what parameters get set in SSML and what in MRCP
and what can be done in both. I don't really like the both solutions.</pre>
</blockquote>
Most of the SSML parameters such as prosody, etc are available in MRCP
as headers. but the values specified in SSML take precendence. The MRCP
header fields allows the client to specify what thedefaultvalues are if
the SSML document doesn't specify it or if the document is not SSML(say
plain text).<br>
<br>
<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
No audioplayer definition.</pre>
</blockquote>
Section 3.2 defines it.<br>
Secition 7.0 describes this. Its essentially a synethsizer that
supports only the &lt;audio&gt; tag in SSML.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 2.2
Doe the user need to be mrcpv2 or is this just an example</pre>
</blockquote>
I don't understand the question. An MRCPv2 server or client is just a
SIP UA which also has some additional functionality.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 3.1, example 1, why bother opening a session before getting any
resources on it. </pre>
</blockquote>
Not required. It was left there to just point out that we could
"possibly" start with 0 channels and then add from there. But there
have been too many questions on this. I will remove it from the next
version, to avoid confusion.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 3.2 - defines  resource type of audioplayer but this is not defined
in the document </pre>
</blockquote>
Well these are synthesizer(Section 7.0) of varying levels of capability
in their SSML support. I think I should put some forward references to
Section 7.0 for these resources. <br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Why did you pick a port of 9 when the port was unknown. Seems like 0 or
2^16-1 would be better?</pre>
</blockquote>
Apparently, thats the NULL port and was recommended for use on the
alias. Zero was my initial choice, but said to be not the right choice.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
If the server is on the inside of a NAT and the client on the outside, can
all of this work? Seems like you want the MRCPv2 control channel to open the
other direction. </pre>
</blockquote>
This may be a problem for the TCP control channel. Suggestions?<br>
<br>
We could allow as an option, for the client to specify the TCP port it
will be listening to (anything other than port 9), in which case the
server needs to open the TCP pipe toqwards the client . Any thoughts?<br>
<br>
There is also another problem that has been brought to my attention,
whose solution may be related.&nbsp; 3PCC of resources.<br>
<br>
When an MRCPv2 client is setting up a pipe between a resource and
another endpoint, It may not have the an initial m=audio line to offer,
may be&nbsp; because it is a part of a 3pcc setup in progress. <br>
So looks like we need to consider a possibility, where the initial
offer for an m=audio line may have to come from the server. This might
require the MRCPv2 client to start with an INVITE to the server with no
offer and wait for the server to provide the first offer. In this case,
we need an alternative mechanism of telling the server, what
resources-types&nbsp; we need, since we&nbsp; would not be allowed to&nbsp; have
m=control lines in the&nbsp; INVITE anyway. <br>
I propose that we allow an XML body of some sort containing list of
resources need, so that&nbsp; server can make the initial offer with the
appropriate&nbsp; m=control lines and m=audio lines in it. <br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Example 3
I'm a little confused on how the listen ports on the TCP stuff work -
particularly when there are multiple of them. If I get two m control lines
that both have 32416 as the listen port at the far side, does that mean I
form two TCP connection to a TCP server listening on that port or I always
use one if it is the same port. If an app was talking to multiple things,
would it need to listen on a bunch of different TCP ports. It might be fine
how this all works but I think a little more text is needed.</pre>
</blockquote>
The protocol just returns the port to talk to open a TCP pipe. between
a client and a server, if there is already one or more TCP pipes open,
they can be shared across multiple MRCP sessions that may be active
between the two.<br>
section 3.2 and 3.4<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 3.3 

Why invent cmid, couldn't you just reuse mid.</pre>
</blockquote>
Each m-line can have an mid. But what we are trying to do here is tie
the m-line with an mid to another m=control line, essentially saying
this media resource should feed from/to that m=audio line.<br>
I cannot use the mid attribute with the same mid in the m=control line
that will fail the mid uniqueness criteria.<br>
The option is the group id defined by the same spec as the mid, that
defines 2 specific group semantics which both don't seem to apply here.
<br>
Do you have something else in mind.<br>
<br>
<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">


Section 5

Why do you need to discuss character sets such as ISO 8859-1. Seems like
saying UTF-8 would be enough.</pre>
</blockquote>
I think ISO 8859 is needed for ASR and TTS content for oriental and
other language support.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
&lt;big issue&gt; It looks like the grammar is set up such there is no easy way to
tell if a start-line is a request-line, event-line, or response-line when
parsing. I think this needs to be fixed either by changing the ABNF or
changing the syntax.</pre>
</blockquote>
Not sure what you mean here.<br>
The request line has a method-name and no state<br>
The response has no method name ans a status and a state.<br>
The event has a method name and a state&nbsp; and no status. <br>
Could you ellaborate.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 5.4
I think line folding is an awful idea, slows down parsers, adds many bugs,
and has no known useful purpose. The fact that other IETF protocols do is
not a known useful purpose.

Allowing both a Header: a,b,c syntax and a multiple Headers of same type was
a bad idea in SIP (see above comment) - I would pick one or the other.
  </pre>
</blockquote>
Need to get back&nbsp; to you on this.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Content-Base: Do you really need to support relative URL at this level? I
understand some specific XML body having a base but can't you just use
absolute URLs at the MRCPv2 level. It's hard to get all the relative stuff
correct particularly when you have ACL based on URL.
  </pre>
</blockquote>
I think this is an optimization. In VXML documents there is a base
URL,&nbsp; which needs to be applied for relative URI references that may
exist within GRXML or SSML documents. The client in many cases does not
parse through the GRXML or SSML documents to expand such cases before
providing to the Server. <br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Content-Encoding: Add that binary MUST be supported and base64 SHOULD NOT be
used. This is a 8bit safe transport.
  </pre>
</blockquote>
fine with me. <br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Logging-tag - I can't decide if I love this or think it is goofy.
  </pre>
</blockquote>
This is a tag that we added in MRCPv1 and also have in VXML so that
logs generated by the application server, VXML browser and the media
processing resources can be corellated.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 7 - point 2 says basicsynth but I think you called it simplesynth in
section 3.2 </pre>
</blockquote>
Will make the correction. I think symplesynth should do.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">

Kill-On-Barge-In - Do you also need some sort of pause-on-barge
  </pre>
</blockquote>
I haven't heard any request for this so far.&nbsp; Any thoughts people.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
When it is a numeric-speech-unit, what is the unit? If undefined, add text
to explain.</pre>
</blockquote>
Dont' understand.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">

page 31 -  Prosody Parameters - do we really need this here if it is also in
SSML.</pre>
</blockquote>
To se the default values, if the SSML does not have the prosody or
other such attributes. Or in cases where the content may not be plain
text.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">

Page 51 
Waveform-URL: What format. There are tons of various formats I can put in
.wav file - I would rather not use file extensions to figure out formats.</pre>
</blockquote>
This was initially added for mostly trouble shooting purposes and to
address the VXMl shadow variable after recognition. This just saves the
utterance or wave form that was used in the recognition. <br>
I don't mind adding headers to specify formats and such for this
recording.<br>
<br>
But note, that there is a separate RECORD method that is being proposed
to do regular recording and stuff that will have mechanisms for the
client to specify storage location, file formats etc.<br>
<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">

DTMF-Term-Char - I don't get this. This is something like # after typing in
a PIN. This and the DTMF inter digit stuff seem like they should be in the
reco grammar not the MRCP.</pre>
</blockquote>
I think this was added to support VXML with DTMF only support, and the
VXML spec allows for such things beyond SSML support. <br>
<h3 id="L8276"><a id="dml6.3.3" name="dml6.3.3" shape="rect">6.3.3
Generic DTMF
Recognizer Properties</a></h3>
of the VXML specification<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">

Page 57
Can you use a cid URL instead of making up the session URL.</pre>
</blockquote>
Well, I thought cid URL was only valid to refer to other parts of a
single multi-part message. <br>
This is a case, where we have defined grammars with prior methods
during the session. These get stored<br>
on the server referenced by the Content-ID that came with content. <br>
Now can we really use the cid to refer that content. I thought this was
not allowed. If thats not the case, I have no problem using cid.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">

Section 8.8 - if a grammar is compiled, how long is it kept
  </pre>
</blockquote>
Wtih the methods in place, grammars defined using section 8.8 are valid
for that session. There are requests for managing server level
grammars, but that has not been addressed yet.<br>
<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">
Section 11 - Is this normative or not. If it conflicts with text which one
is definitive.</pre>
</blockquote>
Section 11 is definitive. Will try to remove the conflicts as much as
possible and make it clear this section is definitive.<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">



Some minor NITS

Lots of formatting issues.

Abstract probably does not need companies that developed MRCPv1.

All the example domain names need to be moved to example.com or such

Section 4 - after defining HX.Y notation, I don't think you use it

Section 6 - bunch of Word got you with " formatting issues.

Section 7 - this would be better drawn as a state machine instead of as a
flow. I am glad to send ASCII art if you want.

Page 32 - top - have SML when I think you mean SSML

Page 34 - "A - value" might be better to say "A negative value"


Page 58 - example 2. Has session URL to local  content but that is not shown
in the example.

Page 59 - reference to NLSML has ref number of 11 but in the bibliography it
is 12.

Section 10
#4 should move to 3261

#8 to #14 have the word " problems

Page 83 - delta-seconds does not seem to be defined


Page 86 numeric-speech-length (and a bunch of others) are redefines of stuff
on page 85</pre>
</blockquote>
<br>
thanks for the comments, I will try to fix them in the next release of
the spec.<br>
<br>
Sarvi<br>
<blockquote type="cite" cite="midBC3527E6.2DC0D%25fluffy@cisco.com">
  <pre wrap="">


Nice work, Easy to read, Thanks, Cullen


  </pre>
</blockquote>
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Date: Tue, 27 Jan 2004 18:13:58 -0800
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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Subject: [Speechsc] MRCPv2 use  in the case of 3pcc - problem
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The current draft does not seem to address the case where the MRCPv2 
client is dealing with a 3pcc INVITE from the endpoint without an 
m=audio line in the offer. But the client knows it wants to bring in a 
Player and Recognizer resource into the session to talk to the endpoint. 
The current call flows seem to have a problem with that.

Has this sort a problem been addressed before.

Option  1:
Can the client just offer m=control lines for the 2 resources with no 
m=audio line, and have the resource side add an m=audio line to the 
answer(i guess that would be more of an offfer with a counter offer or 
answer+offer. I am not sure if this is allowed in the SDP offer/answer 
model.

Option 2:
Send an INVITE with no SDP to the media server,  have the media server 
propose one m=control line for every resource the server supports and 
m=audio line for audio, in the 200 OK. Then have the client in its 
ACK(answer) essentially zero out the ports for the m=control 
lines(resources) it does not need and respond with a port only for the 
resource control line it wants to use.

Option 3:
Send an INVITE (no SDP) with the list of resources it wants to open in 
the code in the URI, the server then proposes m=control line and m=audio 
lines ONLY for the resources requested.

Option 4:
Look for an alternative way(other the URI) for the client to communicate 
that it wants to open X, Y, Z resource at the server so that it can do 
the initial offer. The problem I see here is that, If I wanted to add an 
additional resource later in the session, I don't think SIP allows a 
re-INVITE on the same session with different URI.

Option 5:
Setup only m=control line between the client and the server using SDP 
for all resources and and use an alternative mechanism such as URI or an 
additonal XML or other body to describe the resources to open.

What ever option we pick it needs to work for the case, that we may need 
to later add additional resources on the session, by a re-INVITE or UPDATE.

I am looking for suggestions/comments to address this.

Sarvi




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I would call your example WAY out of scope.

Existing device control choices are J.162 (somewhat documented in =
RFC3435) and H.248.1 (standardized in RFC3525).

Existing application-level choices are VoiceXML (netann) and MSCML =
(documented in mscml).

Do we really need more than that? :-)

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Sunday, January 18, 2004 11:16 AM
> To: SPEECHSC; Eric Burger; Dave Oran
> Subject: Scope of MRCPv2
>=20
>=20
>=20
> I'm trying to figure out the scope of what I should or should=20
> not use MRCPv2
> for so that I can try and decide what might be missing from it.
>=20
> Consider the use case of an IVR build with a separated media=20
> device and
> controller. The media device plays audio prompts, reports=20
> DTMF input up to
> the controller, can stop audio playback when there is DTMF=20
> input, and can
> record audio it receives to files for later playback. It does=20
> not do any
> TTS, ASR, or speaker verification.
>=20
> Is this a use case we want to solve with MRCPv2 or should a different
> protocol be used for this?
>=20
> Thanks, Cullen
>=20
>=20
>=20


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From: Sarvi Shanmugham <sarvi@cisco.com>
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I would tend to disagree here.

The speechrecog resource, in its simplest form can be a DTMF recognizer 
without any changes to the MRCPv2 protocol. Unless ofcourse the question 
was about reporting individual digits as opposed to matching DTMF 
grammars, in which case i agree with you.
The speechsynth resource in its simplest form is an audioplayer(or one 
that can do concatenated prompts at the next level), without any changes 
to the MRCPv2 protocol.
The MRCPv1 extension proposal already has recording support, which 
doesn't seem different from what is requested here.

So I tend to think that the scenario being requested can be addressed 
with spec as is and should meet its needs.
The client may at various times choose to resources of varying levels of 
capabilities such as audioplayer, simplesynth(poor man's TTS) and 
advanced synth or DTMF only recog and speechrecog. The usage semantics 
are thre same and we shouldn't require the client to use different 
protocols for each.

Sarvi

Eric Burger wrote:

>I would call your example WAY out of scope.
>
>Existing device control choices are J.162 (somewhat documented in RFC3435) and H.248.1 (standardized in RFC3525).
>
>Existing application-level choices are VoiceXML (netann) and MSCML (documented in mscml).
>
>Do we really need more than that? :-)
>
>  
>
>>-----Original Message-----
>>From: Cullen Jennings [mailto:fluffy@cisco.com]
>>Sent: Sunday, January 18, 2004 11:16 AM
>>To: SPEECHSC; Eric Burger; Dave Oran
>>Subject: Scope of MRCPv2
>>
>>
>>
>>I'm trying to figure out the scope of what I should or should 
>>not use MRCPv2
>>for so that I can try and decide what might be missing from it.
>>
>>Consider the use case of an IVR build with a separated media 
>>device and
>>controller. The media device plays audio prompts, reports 
>>DTMF input up to
>>the controller, can stop audio playback when there is DTMF 
>>input, and can
>>record audio it receives to files for later playback. It does 
>>not do any
>>TTS, ASR, or speaker verification.
>>
>>Is this a use case we want to solve with MRCPv2 or should a different
>>protocol be used for this?
>>
>>Thanks, Cullen
>>
>>
>>
>>    
>>
>
>
>_______________________________________________
>Speechsc mailing list
>Speechsc@ietf.org
>https://www1.ietf.org/mailman/listinfo/speechsc
>
>  
>

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I would tend to disagree here.<br>
<br>
The speechrecog resource, in its simplest form can be a DTMF recognizer
without any changes to the MRCPv2 protocol. Unless ofcourse the
question was about reporting individual digits as opposed to matching
DTMF grammars, in which case i agree with you.<br>
The speechsynth resource in its simplest form is an audioplayer(or one
that can do concatenated prompts at the next level), without any
changes to the MRCPv2 protocol.<br>
The MRCPv1 extension proposal already has recording support, which
doesn't seem different from what is requested here.<br>
<br>
So I tend to think that the scenario being requested can be addressed
with spec as is and should meet its needs.<br>
The client may at various times choose to resources of varying levels
of capabilities such as audioplayer, simplesynth(poor man's TTS) and
advanced synth or DTMF only recog and speechrecog. The usage semantics
are thre same and we shouldn't require the client to use different
protocols for each.<br>
<br>
Sarvi<br>
<br>
Eric Burger wrote:<br>
<blockquote type="cite"
 cite="mid4A3384433CE2AB46A63468CB207E209DB1A099@zoe.office.snowshore.com">
  <pre wrap="">I would call your example WAY out of scope.

Existing device control choices are J.162 (somewhat documented in RFC3435) and H.248.1 (standardized in RFC3525).

Existing application-level choices are VoiceXML (netann) and MSCML (documented in mscml).

Do we really need more than that? :-)

  </pre>
  <blockquote type="cite">
    <pre wrap="">-----Original Message-----
From: Cullen Jennings [<a class="moz-txt-link-freetext" href="mailto:fluffy@cisco.com">mailto:fluffy@cisco.com</a>]
Sent: Sunday, January 18, 2004 11:16 AM
To: SPEECHSC; Eric Burger; Dave Oran
Subject: Scope of MRCPv2



I'm trying to figure out the scope of what I should or should 
not use MRCPv2
for so that I can try and decide what might be missing from it.

Consider the use case of an IVR build with a separated media 
device and
controller. The media device plays audio prompts, reports 
DTMF input up to
the controller, can stop audio playback when there is DTMF 
input, and can
record audio it receives to files for later playback. It does 
not do any
TTS, ASR, or speaker verification.

Is this a use case we want to solve with MRCPv2 or should a different
protocol be used for this?

Thanks, Cullen



    </pre>
  </blockquote>
  <pre wrap=""><!---->

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Detecting DTMF and playing audio works SO LONG as it is a side-effect, =
and not a goal.  I would not want to see MRCPv2 start handling, for =
example, fax, transcoding, recording for playback purposes, etc.

-----Original Message-----
From: Sarvi Shanmugham [mailto:sarvi@cisco.com]
Sent: Wednesday, January 28, 2004 12:53 PM
To: Eric Burger
Cc: SPEECHSC
Subject: Re: [Speechsc] RE: Scope of MRCPv2


I would tend to disagree here.

The speechrecog resource, in its simplest form can be a DTMF recognizer =
without any changes to the MRCPv2 protocol. Unless ofcourse the question =
was about reporting individual digits as opposed to matching DTMF =
grammars, in which case i agree with you.
The speechsynth resource in its simplest form is an audioplayer(or one =
that can do concatenated prompts at the next level), without any changes =
to the MRCPv2 protocol.
The MRCPv1 extension proposal already has recording support, which =
doesn't seem different from what is requested here.

So I tend to think that the scenario being requested can be addressed =
with spec as is and should meet its needs.
The client may at various times choose to resources of varying levels of =
capabilities such as audioplayer, simplesynth(poor man's TTS) and =
advanced synth or DTMF only recog and speechrecog. The usage semantics =
are thre same and we shouldn't require the client to use different =
protocols for each.

Sarvi

Eric Burger wrote:

I would call your example WAY out of scope.

Existing device control choices are J.162 (somewhat documented in =
RFC3435) and H.248.1 (standardized in RFC3525).

Existing application-level choices are VoiceXML (netann) and MSCML =
(documented in mscml).

Do we really need more than that? :-)

 =20
-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: Sunday, January 18, 2004 11:16 AM
To: SPEECHSC; Eric Burger; Dave Oran
Subject: Scope of MRCPv2



I'm trying to figure out the scope of what I should or should=20
not use MRCPv2
for so that I can try and decide what might be missing from it.

Consider the use case of an IVR build with a separated media=20
device and
controller. The media device plays audio prompts, reports=20
DTMF input up to
the controller, can stop audio playback when there is DTMF=20
input, and can
record audio it receives to files for later playback. It does=20
not do any
TTS, ASR, or speaker verification.

Is this a use case we want to solve with MRCPv2 or should a different
protocol be used for this?

Thanks, Cullen



   =20


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

 =20


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Date: Thu, 29 Jan 2004 12:05:48 -0800
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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Though DTMF was not the primary goal,  it is supported through DTMF only 
headers like Dtmf-Term-Char etc. that were added to support VXML 
documents with DTMF only grammars.
 I don't believe we have done anything special for audio play back, its 
just a synthesizer with limited capabilities.
There is a need for recording as part of MRCP that has been voiced 
earlier and agreed on during the last meeting. The MRCPv1 extension 
proposes a Recording capability where the audio can be played back. A 
few reasons why recording functionality is being added is that in some 
cases,
    1. we need functionality such as audio end-pointing(related to 
recognizers) to start and stop recording.
    2. we may need to co-ordinate with DTMF recognition and play back 
for starting and stoping recordings.
And again like you say, it was added mostly because there were some 
requirements related to recognizers and synthesizers. But I believe we 
should limit its use  if some one wants to use it as plain recorder for 
recording voice mail or something.

Fax and Transcoding I agree are not it scope.

Sarvi


Eric Burger wrote:

>Detecting DTMF and playing audio works SO LONG as it is a side-effect, and not a goal.  I would not want to see MRCPv2 start handling, for example, fax, transcoding, recording for playback purposes, etc.
>
>-----Original Message-----
>From: Sarvi Shanmugham [mailto:sarvi@cisco.com]
>Sent: Wednesday, January 28, 2004 12:53 PM
>To: Eric Burger
>Cc: SPEECHSC
>Subject: Re: [Speechsc] RE: Scope of MRCPv2
>
>
>I would tend to disagree here.
>
>The speechrecog resource, in its simplest form can be a DTMF recognizer without any changes to the MRCPv2 protocol. Unless ofcourse the question was about reporting individual digits as opposed to matching DTMF grammars, in which case i agree with you.
>The speechsynth resource in its simplest form is an audioplayer(or one that can do concatenated prompts at the next level), without any changes to the MRCPv2 protocol.
>The MRCPv1 extension proposal already has recording support, which doesn't seem different from what is requested here.
>
>So I tend to think that the scenario being requested can be addressed with spec as is and should meet its needs.
>The client may at various times choose to resources of varying levels of capabilities such as audioplayer, simplesynth(poor man's TTS) and advanced synth or DTMF only recog and speechrecog. The usage semantics are thre same and we shouldn't require the client to use different protocols for each.
>
>Sarvi
>
>Eric Burger wrote:
>
>I would call your example WAY out of scope.
>
>Existing device control choices are J.162 (somewhat documented in RFC3435) and H.248.1 (standardized in RFC3525).
>
>Existing application-level choices are VoiceXML (netann) and MSCML (documented in mscml).
>
>Do we really need more than that? :-)
>
>  
>-----Original Message-----
>From: Cullen Jennings [mailto:fluffy@cisco.com]
>Sent: Sunday, January 18, 2004 11:16 AM
>To: SPEECHSC; Eric Burger; Dave Oran
>Subject: Scope of MRCPv2
>
>
>
>I'm trying to figure out the scope of what I should or should 
>not use MRCPv2
>for so that I can try and decide what might be missing from it.
>
>Consider the use case of an IVR build with a separated media 
>device and
>controller. The media device plays audio prompts, reports 
>DTMF input up to
>the controller, can stop audio playback when there is DTMF 
>input, and can
>record audio it receives to files for later playback. It does 
>not do any
>TTS, ASR, or speaker verification.
>
>Is this a use case we want to solve with MRCPv2 or should a different
>protocol be used for this?
>
>Thanks, Cullen
>
>
>
>    
>
>
>_______________________________________________
>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		: Requirements for Distributed Control of ASR, SI/SV and TTS Resources
	Author(s)	: D. Oran
	Filename	: draft-ietf-speechsc-reqts-05.txt
	Pages		: 18
	Date		: 2004-1-29
	
This document outlines the needs and requirements for a protocol to
control distributed speech processing of audio streams.  By speech
processing, this document specifically means automatic speech
recognition (ASR), speaker recognition - which includes both speaker
identification (SI) and speaker verification (SV) - and
text-to-speech (TTS).  Other IETF protocols, such as SIP and RTSP,
address rendezvous and control for generalized media streams.
However, speech processing presents additional requirements that none
of the extant IETF protocols address.

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Channel Identifier (S 5.4)

Why is the channel identifier hex?  Why is it not just a token -- a =
handle passed back and forth?

Also, the text still says the low two characters indicate the resource, =
but we have the @resource construction that makes the encoding =
redundant.

Is there any reason for this being a number and not a string?  If it is =
a string, I could see it being case sensitive to expand the token space, =
or leave it case insensitive, as there is no limit to the length of the =
string.

Should there be a length of the string?



Proxy-Sync-Id, Logging-Tag
Same thing here.  Why is it a letter string and not a token?  This =
really is a cookie.


Recognizer Context Block
Is this always ASCII printable A-Z characters?  Does the server =
community need this to be a binary object?


Failed URI Cause
Is this supposed to be the code and text from the protocol, e.g., if =
http?  If so, the ABNF does not allow spaces.

Input from the server community welcome, too.


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Thread-Topic: mrcpv2: Audio formats
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Subject: [Speechsc] mrcpv2: Audio formats
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The current text says that .wav and .au be supported for audio playback.

Cullen is correct in stating that .wav files do not limit the codec.

What if we say that .au MUST be supported, and .wav MUST be supported =
with G.711 u-Law and A-Law MUST be supported?

Same issue on recording (if we do recording).


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Subject: [Speechsc] mrcpv2-01: Media Streams (S. 3.3)
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Section 3.3 talks about the behavior and suggested directionality for =
media streams when one hooks up players and recognizers.

What happens when someone gets it wrong?  For example, what if I set up =
a player with sendonly (remembering that we're talking about =
directionality from the client)?  Do I accept the client's SDP, figuring =
they know what they're doing, and will not be surprised when they never =
receive audio?  Or do I reject the offer with some sort of SIP 'bad SIP' =
response?

Opinions?


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Subject: [Speechsc] mrcpv2-01: Strength of BARGE-IN-OCCURED (S 7.10) PAUSE (S 7.11) and RESUME (S 7.12)
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Is there a situation where the receipt of a BARGE-IN-OCCURED request =
while kill-on-barge-in is enabled would *not* stop streaming audio?  If =
not, then the strength of stopping audio must be MUST :-)

Same question for PAUSE - if something is playing and I PAUSE it, is =
there a reason not to pause it?

Same question for RESUME - if something is paused and I RESUME it, is =
there a reason not to resume it?

What else would we do?  If nothing, then change the strength to MUST.


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Thread-Topic: mrcpv2-01: Behavior of CONTROL
Thread-Index: AcPm64BVln8F8x1tRPSydFRe6I9xIA==
From: "Eric Burger" <eburger@snowshore.com>
To: "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
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Subject: [Speechsc] mrcpv2-01: Behavior of CONTROL
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The behavior described is sensible.  However, the use case presented is =
not satisfied by the behavior.

My interpretation of the text is that if you forward past the end of the =
current object, the current object stops play and the completion event =
is generated.  What if there are queued objects to play?  My read of the =
text is that they will start playing immediately, *from the beginning of =
the object*.

Likewise, if you rewind past the beginning of an object, that object =
will start playback at the beginning of the object, with the =
speech-restart header (flag) present.

Neither of these behaviors allow rewind or skip-ahead across multiple =
speech requests.  The client has to do it manually.

I actually like the behavior described above, as the client can figure =
out where to skip to, so long as the completion event and speech-restart =
has information as to how much time was left.

That said, is it meaningful to say "skip ahead 2 seconds while playing =
'a long sentence going through a TTS resource that runs in real time so =
it has no idea where 2 seconds is'"?


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Thread-Topic: mrcpv2-01: Strength of OPTIONS processing (S 6)
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Subject: [Speechsc] mrcpv2-01: Strength of OPTIONS processing (S 6)
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Does it make sense for the result of an OPTIONS to *not* have at least =
the m-line for the control channel?  Is there a situation where a server =
does *not* support at least one media type?  If not, my vote is the =
server MUST return the appropriate m-lines in response to the OPTIONS =
request.


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MUST the recognizer store the utterance?  I would think this should be a =
SHOULD.  What if the recognizer does not have storage?  What if it does =
have storage but it is full?  What if it is in overload and does not do =
storage processing?

Thoughts?


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Thread-Topic: mrcpv2-01: Security Considerations and URI's
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Subject: [Speechsc] mrcpv2-01: Security Considerations and URI's
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The document needs a security considerations section.

The protocol allows for fetching by reference.  We need to point out the =
potential for attack amplification (one MRCPv2 request generating lots =
of fetches) -- both on the server itself and as a platform for a DDoS =
attack.

How do we want to handle URI's.  For interoperability, we should mandate =
certain schemas.  I would vote for http being a MUST implement and https =
being a SHOULD.  Is the requirement for http too strong?

What happens if the server does not support the URI scheme?  E.g., a =
file://mumble/foo or ftp://mumble/foo?  What if the scheme doesn't make =
sense, e.g., tel:3211234567?


URI's appear in the Synthesizer Speech Data fetch list (S 7.5), Waveform =
URL (S 8.4)


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Thread-Topic: mrcpv2-01: Speech Language (S 8.4)
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Is there a reason to have any other language specification other than =
RFC1766 (e.g., MUST strength, if present)?


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Right now it is a free-for-all to have some kind of internal tagging for =
the recognizer context block to let a vendor know that the block belongs =
to them.

How about adding a vendor tag?  We can have an IANA registry of vendor =
tag names, so we can be sensible (e.g., no x-mumble).


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How about the DTMF detector MUST support RFC2833 and MAY detect in-band =
DTMF tones?  Speech recognition detector MAY support DTMF.  If it does, =
it MUST support RFC2833 and MAY detect in-band DTMF.

Sound OK, or is this something that really is up to the ASR vendors to =
decide?


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One the one hand, the text says Content-Length MUST be present.  On the =
other hand, the text says if it is not present, it is OK.

Which should it be?  Choose one.  Given the number of parameters, I =
could see rejecting a message without a Content-Length.  Otherwise, my =
vote is to drop the MUST.


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