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Subject: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-07.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)	: D. Burnett, S. Shanmugham
	Filename	: draft-ietf-speechsc-mrcpv2-07.txt
	Pages		: 198
	Date		: 2005-8-1
	
The MRCPv2 protocol allows client hosts to control media service
   resources such as speech synthesizers, recognizers, verifiers and
   identifiers residing in servers on the network.  MRCPv2 is not a
   "stand-alone" protocol - it relies on a session management protocol
   such as the Session Initiation Protocol (SIP) to establish the MRCPv2
   control session between the client and the server, and for rendezvous
   and capability discovery.  It also depends on SIP and SDP to
   establish the media sessions and associated parameters between the
   media source or sink and the media server.  Once this is done, the
   MRCPv2 protocol exchange operates over the control session
   established above, allowing the client to control the media
   processing resources on the speech resource server.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-07.txt

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From speechsc-bounces@ietf.org Thu Aug 04 13:43:43 2005
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Subject: RE: [Speechsc] Server capabilities
Date: Thu, 4 Aug 2005 10:43:28 -0700
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Thread-Topic: [Speechsc] Server capabilities
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From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Corby Anderson" <corby@tellme.com>, <Speechsc@ietf.org>
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Here is a suggestion to address this issue.

The ability to discover the capabilities of an MRCP server is
substantial piece of work in my opnion. Something that need a draft of
its own. Its not a priority issue that should stop the MRCPv2 draft from
progressing. I would suggest that we defer this work to a separate draft
and allow us to finish this one. The current draft is pretty huge by
itself.
  I like the idea of using SIP OPTIONS along with necessary SDP
definitions that describe the capabilities of resources it supports.
These I believe can be defered to a separate draft without affecting
MRCPv2 very much.=20
  This draft should also define mime-bodies that cover support for
certain headers and their possible values(like CONTROL method and the
different values supported for JumpSize etc).=20
  It should also return a mime-body listing the grammars that are
provisioned permanently on the server.

What we not should do.
    I don't believe the installation of grammars that may be persistent
across sessions should be the job of MRCPv2. I believe this should be a
provisioning issue and addressed through SNMP or some other approach
that may be used to install the server, the different
resources/languages it supports etc.

Sarvi =20

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Shanmugham, Saravanan
     Sent: Wednesday, June 15, 2005 12:49 PM
     To: Corby Anderson; Speechsc@ietf.org
     Subject: RE: [Speechsc] Server capabilities
    =20
     This comment is only relating to your suggestion to use=20
     option b to address resolving binary grammar uri.
     I think the right way to do this would be to define a file=20
     format for this binary across vendors and use it. This=20
     wouldn't have to be very complex, at a minimum agree on=20
     standard header for the beginning of this file that=20
     identifies vendor and format version number and the rest=20
     of the binary file could be vendor specific.=20
    =20
     No need to address this through the MRCP specification.
    =20
     Sarvi=20
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org=20
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Corby Anderson
          Sent: Wednesday, June 15, 2005 12:20 PM
          To: Speechsc@ietf.org
          Subject: Re: [Speechsc] Server capabilities
         =20
          I'm very much in favor of b).
         =20
          In the "managing big grammars" thread we were considering=20
          about permitting URIs to reference binary grammars as well=20
          as source grammars (was that ever codified?).  How about=20
          having some way for the server to indicate what=20
          vendor/version it is.  For heterogenous MRCP server=20
          environments, this would allow clients to give the right=20
          grammar binary URIs to the right servers.
         =20
          Corby Anderson
          Tellme Networks, Inc.
         =20
          David R. Oran wrote:
         =20
          > This message was prompted by one aspect of the=20
          discussion on the list=20
          > about large grammars, which is how a client can figure=20
          out in advance=20
          > just what a server can and can't do, so it isn't=20
     "unpleasantly=20
          > surprised" at the instant it issues a method.
          >
          > The current specification walks a fine line between=20
          giving flexibility=20
          > to server implementers and giving clients a=20
     sufficient degree of=20
          > control over server behavior. For example we have things=20
          like multiple=20
          > jump-length units which a server may or may not support,=20
          > server-specific default values for confidence-threshold and=20
          > speed-vs-accuracy, and a bunch of others.
          >
          > There is some for this support, including:
          > 1) through SDP and SIP OPTIONS the client can discover=20
          what resources=20
          > the server supports and what media formats it can process.
          > 2) through GET-PARAMS on a particular session with a=20
          resource the=20
          > client can discover any implementation-specific server=20
          default values=20
          > for MRCPv2 protocol headers.
          >
          > My question to the WG is what, if anything do we need=20
          beyond these?
          >
          > Here's a small list of candidates. Feel free to comment=20
          and/or propose=20
          > your own:
          >
          > a) we may want/need a way for the server to=20
     determine what audio=20
          > formats a client supports for the purposes of=20
     returning captured=20
          > audio. Right now we don't even have a way for the client=20
          to tell the=20
          > server what format it wants audio returned in, so we at=20
          least have to=20
          > remedy that omission.
          >
          > b) we may want a way for a client to learn the=20
     current list of=20
          > "instantly available" grammars the server has handy and=20
          which are in=20
          > some way "guaranteed" to persist for a while.
          >
          > c) we may want a form of GET-PARAMS which fetches=20
          session-independent=20
          > server information. One way to do this is to define a=20
          MRCPv2-frag=20
          > content type (like we have with sipfrag). That way a SIP=20
          OPTIONs to an=20
          > MRCPv2 server can return an MRCP-specific body with=20
     all that=20
          > interesting information about server capabilities.
          >
          > <chair hat>
          > I don't want this to necessary turn into a free-for-all,=20
          because we=20
          > really do want to reach closure on the spec and get it=20
          to last call.
          > My target for finishing my chair review and editing is=20
          the end of this=20
          > week.
          > </chair hat>
          >
          > _______________________________________________
          > Speechsc mailing list
          > Speechsc@ietf.org
          > https://www1.ietf.org/mailman/listinfo/speechsc
         =20
         =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
         =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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



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Subject: [Speechsc] Driving toward Last Call on MRCPv2
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By now you folks should have noticed that -07 was posted by Sarvi.  
This was a major revision, incorporating resolutions to the vast  
majority of the open issues, a conversion from Word to XML source,  
and a comprehensive technical and editorial review by both co-chairs.

If you haven't already, you can get it from:
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-07.txt

In order to keep track of the issues raised on the mailing list and  
to manage the editing process, the chairs and authors have been using  
the "roundup" issue tracking tool. It is being hosted on  
softarmor.com with the gracious help of Dean Willis. We kept it  
private for the creation of -07, but are now ready to open it to the  
whole working group to review the changes and see the status of the  
remaining open issues.

Please navigate to https://www.softarmor.com/roundup/speechsc/ and  
register yourself to access the issues list.

The chairs would like to have no more than one further revision of  
the specification and be ready for WG last call by the end of August.  
Therefore, please:
- make sure any issues you have reaised and were supposed to be  
addressed in -07 in fact have been marked resolved in the issue  
tracker and correctly reflected in -07
- review the open issues and indicate on the mailing list if there  
you feel their are any which MUST be resolved prior to going to last  
call. Unless we hear otherwise, we would like to defer non-critical  
functionality from this version of MRCP so we can get to proposed  
standard soon.

We would like to declare rough consensus on each of the open issues  
as soon as feasible, and surface any new issues and deal with them  
expeditiously.

I know this is mid-summer and people have lots of other things to  
think about. Some of your time and energy on this effort will be of  
great assistance to the working group in completing our main piece of  
technical work with minimal further delay.

Thanks and Regards,
Dave Oran (speechsc co-chair).


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



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What's the status of this request as it doesn't appear to have been 
addressed in the 07 draft or the roundup tool?

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/ 
gavagni@us.ibm.com




"Dave Burke" <david.burke@voxpilot.com> 
Sent by: speechsc-bounces@ietf.org
06/01/2005 11:30 AM

To
"David R. Oran" <oran@cisco.com>, "Reifenrath, Klaus" 
<Klaus.Reifenrath@Scansoft.com>
cc
speechsc@ietf.org, Jerry Carter <jerry@jerrycarter.org>
Subject
Re: [Speechsc] EMMA: Extensible MultiModal Annotation markup language 
support?






Having experienced the pain of several MRCPv1 ASR integrations (almost 
exclusively because of the previously insufficiently defined reco 
results), 
we agree with the comment for "stable base for development and 
deployment".

Worried that even a "MAY support EMMA" is damaging but think it a 
reasonable 
compromise to add an informative note pointing out that the intent is to 
incorporate EMMA when it is a stable standard. This way the direction is 
clear and we don't have to revisit this question bimonthly!

-- Dave

----- Original Message ----- 
From: "David R. Oran" <oran@cisco.com>
To: "Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>
Cc: <speechsc@ietf.org>; "Jerry Carter" <jerry@jerrycarter.org>
Sent: Wednesday, June 01, 2005 3:07 PM
Subject: Re: [Speechsc] EMMA: Extensible MultiModal Annotation markup 
language support?


> **Chair hat on**
>
> The WG consensus, established over a number of IETF meetings and mailing 

> list discussion, was to publish MRCPv2 with the embedded  NLSML so that 
> implementations would have a stable base for  development and initial 
> deployment.
>
> The intent has always been to incorporate EMMA once it was deemed  fully 

> cooked.
>
> If the WG wants to revisit this consensus in light of the fact we  will 
be 
> going to last call on MRCPv2 imminently, I'll certainly  entertain that 
> possibility.
>
> Dave.
>
>
> On Jun 1, 2005, at 9:50 AM, Reifenrath, Klaus wrote:
>
>> Hi Brett,
>>
>> right, NLSML did not become a W3C Recommendation. Therefore the  MRCPv2 

>> spec
>> includes a definition of NLSML (see sections 9.6 and A.2.1 in the  06 
>> version
>> of MRCPv2).
>>
>> Klaus
>>
>> -----Original Message-----
>> From: Brett Gavagni [mailto:gavagni@us.ibm.com]
>> Sent: Mittwoch, 1. Juni 2005 15:34
>> To: Jerry Carter
>> Cc: speechsc@ietf.org
>> Subject: Re: [Speechsc] EMMA: Extensible MultiModal Annotation markup
>> language support?
>>
>> The current W3 NLSML specification is also a "W3 Working Draft" and not 

>> a
>> "W3 Recommendation".
>>
>> http://www.w3.org/TR/nl-spec/
>> Natural Language Semantics Markup Language for the Speech Interface
>> Framework
>>
>> W3C Working Draft 20 November 2000
>>
>> The current NLSML specification is actually more outdated in terms  of 
>> the
>> function required to support the current SGRS specification (ie. 
>> Semantic
>> Interpretation).
>>
>> My suggestion would be to amend the current MRCPv2 specification to 
>> include
>> EMMA support as at least optional.
>>
>> Thanks,
>>
>> Brett Gavagni
>> WebSphere Voice Server Development
>> http://www-306.ibm.com/software/pervasive/voice_server/
>> (561) 862-2097 T/L (975)
>> gavagni@us.ibm.com
>>
>>
>>
>>
>> Jerry Carter <jerry@jerrycarter.org>
>> 06/01/2005 09:16 AM
>>
>> To
>> Brett Gavagni/West Palm Beach/IBM@IBMUS
>> cc
>> speechsc@ietf.org
>> Subject
>> Re: [Speechsc] EMMA: Extensible MultiModal Annotation markup language
>> support?
>>
>>
>>
>>
>>
>>
>> Give that EMMA is at working draft level within the W3C, it would be
>> premature to require any form of EMMA support.  Making a provision in
>> MRCP to support EMMA at a later date would be appropriate, though.
>>
>> On Jun 1, 2005, at 8:59 AM, Brett Gavagni wrote:
>>
>>
>>> Are there currently any plans to include the support for W3C EMMA
>>> Working
>>> Draft to the MRCPv2 specification for the returning of recognition
>>> results?
>>>
>>> Thanks,
>>>
>>> Brett Gavagni
>>> WebSphere Voice Server Development
>>> http://www-306.ibm.com/software/pervasive/voice_server/
>>> (561) 862-2097 T/L (975)
>>> gavagni@us.ibm.com
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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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Subject: Re: [Speechsc] Personal-Grammar-URI clarification
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Bump

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/ 
gavagni@us.ibm.com




Brett Gavagni/West Palm Beach/IBM@IBMUS 
Sent by: speechsc-bounces@ietf.org
07/26/2005 11:28 AM

To
speechsc@ietf.org
cc

Subject
[Speechsc] Personal-Grammar-URI clarification






Hi,

The wording throughout the current draft specification is a bit confusing 
w.r.t "Personal-Grammar-URI".

Is the expectation that the "Personal-Grammar-URI" value is merely a 
client specified handle to an external grammar in lieu of using a 
"Content-Id" value?

Is the expectation that the server persistently store these references and 

the backing grammars, and when and how would the release of these 
references occur?

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/ 
gavagni@us.ibm.com


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



_______________________________________________
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From speechsc-bounces@ietf.org Wed Aug 10 14:55:18 2005
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Subject: RE: [Speechsc] EMMA: Extensible MultiModal Annotation markup
	languagesupport?
Date: Wed, 10 Aug 2005 11:55:03 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C3902@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] EMMA: Extensible MultiModal Annotation markup
	languagesupport?
Thread-Index: AcWd2zoK+6orY+o8T9a0hqQrN9oTRQAAbScg
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Brett Gavagni" <gavagni@us.ibm.com>,
	"Dave Burke" <david.burke@voxpilot.com>
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	speechsc-bounces@ietf.org, Jerry Carter <jerry@jerrycarter.org>,
	"Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>
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Added to this list. The text proposed by Dave will be added to the next
draft.

Sarvi=20

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
     Sent: Wednesday, August 10, 2005 11:36 AM
     To: Dave Burke
     Cc: speechsc@ietf.org; David R. Oran;=20
     speechsc-bounces@ietf.org; Jerry Carter; Reifenrath, Klaus
     Subject: Re: [Speechsc] EMMA: Extensible MultiModal=20
     Annotation markup languagesupport?
    =20
     What's the status of this request as it doesn't appear to=20
     have been addressed in the 07 draft or the roundup tool?
    =20
     Thanks,
    =20
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
    =20
    =20
    =20
    =20
     "Dave Burke" <david.burke@voxpilot.com> Sent by:=20
     speechsc-bounces@ietf.org
     06/01/2005 11:30 AM
    =20
     To
     "David R. Oran" <oran@cisco.com>, "Reifenrath, Klaus"=20
     <Klaus.Reifenrath@Scansoft.com>
     cc
     speechsc@ietf.org, Jerry Carter <jerry@jerrycarter.org> Subject
     Re: [Speechsc] EMMA: Extensible MultiModal Annotation=20
     markup language support?
    =20
    =20
    =20
    =20
    =20
    =20
     Having experienced the pain of several MRCPv1 ASR=20
     integrations (almost=20
     exclusively because of the previously insufficiently defined reco=20
     results),=20
     we agree with the comment for "stable base for development and=20
     deployment".
    =20
     Worried that even a "MAY support EMMA" is damaging but think it a=20
     reasonable=20
     compromise to add an informative note pointing out that=20
     the intent is to=20
     incorporate EMMA when it is a stable standard. This way=20
     the direction is=20
     clear and we don't have to revisit this question bimonthly!
    =20
     -- Dave
    =20
     ----- Original Message -----=20
     From: "David R. Oran" <oran@cisco.com>
     To: "Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>
     Cc: <speechsc@ietf.org>; "Jerry Carter" <jerry@jerrycarter.org>
     Sent: Wednesday, June 01, 2005 3:07 PM
     Subject: Re: [Speechsc] EMMA: Extensible MultiModal=20
     Annotation markup=20
     language support?
    =20
    =20
     > **Chair hat on**
     >
     > The WG consensus, established over a number of IETF=20
     meetings and mailing=20
    =20
     > list discussion, was to publish MRCPv2 with the embedded=20
      NLSML so that=20
     > implementations would have a stable base for =20
     development and initial=20
     > deployment.
     >
     > The intent has always been to incorporate EMMA once it=20
     was deemed  fully=20
    =20
     > cooked.
     >
     > If the WG wants to revisit this consensus in light of=20
     the fact we  will=20
     be=20
     > going to last call on MRCPv2 imminently, I'll certainly =20
     entertain that=20
     > possibility.
     >
     > Dave.
     >
     >
     > On Jun 1, 2005, at 9:50 AM, Reifenrath, Klaus wrote:
     >
     >> Hi Brett,
     >>
     >> right, NLSML did not become a W3C Recommendation.=20
     Therefore the  MRCPv2=20
    =20
     >> spec
     >> includes a definition of NLSML (see sections 9.6 and=20
     A.2.1 in the  06=20
     >> version
     >> of MRCPv2).
     >>
     >> Klaus
     >>
     >> -----Original Message-----
     >> From: Brett Gavagni [mailto:gavagni@us.ibm.com]
     >> Sent: Mittwoch, 1. Juni 2005 15:34
     >> To: Jerry Carter
     >> Cc: speechsc@ietf.org
     >> Subject: Re: [Speechsc] EMMA: Extensible MultiModal=20
     Annotation markup
     >> language support?
     >>
     >> The current W3 NLSML specification is also a "W3=20
     Working Draft" and not=20
    =20
     >> a
     >> "W3 Recommendation".
     >>
     >> http://www.w3.org/TR/nl-spec/
     >> Natural Language Semantics Markup Language for the=20
     Speech Interface
     >> Framework
     >>
     >> W3C Working Draft 20 November 2000
     >>
     >> The current NLSML specification is actually more=20
     outdated in terms  of=20
     >> the
     >> function required to support the current SGRS=20
     specification (ie.=20
     >> Semantic
     >> Interpretation).
     >>
     >> My suggestion would be to amend the current MRCPv2=20
     specification to=20
     >> include
     >> EMMA support as at least optional.
     >>
     >> Thanks,
     >>
     >> Brett Gavagni
     >> WebSphere Voice Server Development
     >> http://www-306.ibm.com/software/pervasive/voice_server/
     >> (561) 862-2097 T/L (975)
     >> gavagni@us.ibm.com
     >>
     >>
     >>
     >>
     >> Jerry Carter <jerry@jerrycarter.org>
     >> 06/01/2005 09:16 AM
     >>
     >> To
     >> Brett Gavagni/West Palm Beach/IBM@IBMUS
     >> cc
     >> speechsc@ietf.org
     >> Subject
     >> Re: [Speechsc] EMMA: Extensible MultiModal Annotation=20
     markup language
     >> support?
     >>
     >>
     >>
     >>
     >>
     >>
     >> Give that EMMA is at working draft level within the=20
     W3C, it would be
     >> premature to require any form of EMMA support.  Making=20
     a provision in
     >> MRCP to support EMMA at a later date would be=20
     appropriate, though.
     >>
     >> On Jun 1, 2005, at 8:59 AM, Brett Gavagni wrote:
     >>
     >>
     >>> Are there currently any plans to include the support=20
     for W3C EMMA
     >>> Working
     >>> Draft to the MRCPv2 specification for the returning of=20
     recognition
     >>> results?
     >>>
     >>> Thanks,
     >>>
     >>> Brett Gavagni
     >>> WebSphere Voice Server Development
     >>> http://www-306.ibm.com/software/pervasive/voice_server/
     >>> (561) 862-2097 T/L (975)
     >>> gavagni@us.ibm.com
     >>>
     >>>
     >>> _______________________________________________
     >>> 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
     >>
     >> _______________________________________________
     >> 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
     >=20
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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Subject: RE: [Speechsc] Personal-Grammar-URI clarification
Date: Wed, 10 Aug 2005 13:03:41 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C392A@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] Personal-Grammar-URI clarification
Thread-Index: AcWd26R/Ig6cKD/STkyx3yPzjS/YcQAAWWvw
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
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I am not sure what you mean by its relationship to Content-Id.

To me Personal-Grammar-URI is just like anyother grammar that has a
<one-of> list of items.=20
Except that it is constructed by the user adding individual phrases to
that one-of list during enrollment.  Once enrolled, the usage of that
grammar should be no different from any other grammar.

This could be a HTTP URI in which case the MRCP server uses HTTP GET/PUT
to modify that grammar during enrollment. So to answer your question,
the MRCP server is not expected to store this personal grammar, but
would be create/modified at the URI specified by the personal grammar
URI.

Sarvi

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
     Sent: Wednesday, August 10, 2005 11:43 AM
     To: speechsc@ietf.org
     Subject: Re: [Speechsc] Personal-Grammar-URI clarification
    =20
     Bump
    =20
     Thanks,
    =20
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
    =20
    =20
    =20
    =20
     Brett Gavagni/West Palm Beach/IBM@IBMUS Sent by:=20
     speechsc-bounces@ietf.org
     07/26/2005 11:28 AM
    =20
     To
     speechsc@ietf.org
     cc
    =20
     Subject
     [Speechsc] Personal-Grammar-URI clarification
    =20
    =20
    =20
    =20
    =20
    =20
     Hi,
    =20
     The wording throughout the current draft specification is=20
     a bit confusing w.r.t "Personal-Grammar-URI".
    =20
     Is the expectation that the "Personal-Grammar-URI" value=20
     is merely a client specified handle to an external grammar=20
     in lieu of using a "Content-Id" value?
    =20
     Is the expectation that the server persistently store=20
     these references and=20
    =20
     the backing grammars, and when and how would the release=20
     of these references occur?
    =20
     Thanks,
    =20
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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Subject: RE: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-07.txt 
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When is the MRCP draft document going to be revised with the synthesizer
and recognizer usage tables included?  The following text in draft 07
needs to be replaced with the actual table (same for the recognizer
resource):

   SYNTHESIZER HEADER USAGE TABLE TEMPORARILY REMOVED DUE TO XML2RFC
   LAYOUT PROBLEMS

Thanks,
Pierre

-----Original Message-----
From: speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] On
Behalf Of Internet-Drafts@ietf.org
Sent: Monday, August 01, 2005 3:50 PM
To: i-d-announce@ietf.org
Cc: speechsc@ietf.org
Subject: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-07.txt=20

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)	: D. Burnett, S. Shanmugham
	Filename	: draft-ietf-speechsc-mrcpv2-07.txt
	Pages		: 198
	Date		: 2005-8-1
=09
The MRCPv2 protocol allows client hosts to control media service
   resources such as speech synthesizers, recognizers, verifiers and
   identifiers residing in servers on the network.  MRCPv2 is not a
   "stand-alone" protocol - it relies on a session management protocol
   such as the Session Initiation Protocol (SIP) to establish the MRCPv2
   control session between the client and the server, and for rendezvous
   and capability discovery.  It also depends on SIP and SDP to
   establish the media sessions and associated parameters between the
   media source or sink and the media server.  Once this is done, the
   MRCPv2 protocol exchange operates over the control session
   established above, allowing the client to control the media
   processing resources on the speech resource server.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-07.txt

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On Aug 17, 2005, at 10:36 AM, Pierre Forgues wrote:

> When is the MRCP draft document going to be revised with the  
> synthesizer
> and recognizer usage tables included?  The following text in draft 07
> needs to be replaced with the actual table (same for the recognizer
> resource):
>
>    SYNTHESIZER HEADER USAGE TABLE TEMPORARILY REMOVED DUE TO XML2RFC
>    LAYOUT PROBLEMS
>
Maybe never, if we can't figure out a way to coax XML2RFC into  
meeting the 72-character line limit imposed by the RFC-editor, or  
figure out a way to re-layout the tables in a way that makes them fit.

How'd you like to take a crack at this? I already wasted a number of  
hours.

BTW: In the case of other specs (e.g. SIP) the community has come to  
the tentative conclusion that the tables are a maintenance/ 
consistency nightmare and might be better left out in the first  
place. It's an option we should consider...

Dave.


> Thanks,
> Pierre
>
> -----Original Message-----
> From: speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] On
> Behalf Of Internet-Drafts@ietf.org
> Sent: Monday, August 01, 2005 3:50 PM
> To: i-d-announce@ietf.org
> Cc: speechsc@ietf.org
> Subject: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-07.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)    : D. Burnett, S. Shanmugham
>     Filename    : draft-ietf-speechsc-mrcpv2-07.txt
>     Pages        : 198
>     Date        : 2005-8-1
>
> The MRCPv2 protocol allows client hosts to control media service
>    resources such as speech synthesizers, recognizers, verifiers and
>    identifiers residing in servers on the network.  MRCPv2 is not a
>    "stand-alone" protocol - it relies on a session management protocol
>    such as the Session Initiation Protocol (SIP) to establish the  
> MRCPv2
>    control session between the client and the server, and for  
> rendezvous
>    and capability discovery.  It also depends on SIP and SDP to
>    establish the media sessions and associated parameters between the
>    media source or sink and the media server.  Once this is done, the
>    MRCPv2 protocol exchange operates over the control session
>    established above, allowing the client to control the media
>    processing resources on the speech resource server.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-07.txt
>
> To remove yourself from the I-D Announcement list, send a message to
> i-d-announce-request@ietf.org with the word unsubscribe in the body of
> the message.
> You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce
> to change your subscription settings.
>
>
> Internet-Drafts are also available by anonymous FTP. Login with the
> username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>     "get draft-ietf-speechsc-mrcpv2-07.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-mrcpv2-07.txt".
>
> NOTE:    The mail server at ietf.org can return the document in
>     MIME-encoded form by using the "mpack" utility.  To use this
>     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
>

_______________________________________________
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To: "Shanmugham, Saravanan" <sarvi@cisco.com>
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Thanks for the clarification!

Since SRGS defines syntactical support for both ABNF and XML formats, it 
may be useful to add some wording or modify the "Personal-Grammar-URI" 
examples to specify the "Content-Type".

This may assist in facilitating client/server interoperability. 

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/
gavagni@us.ibm.com




"Shanmugham, Saravanan" <sarvi@cisco.com> 
08/10/2005 04:03 PM

To
Brett Gavagni/West Palm Beach/IBM@IBMUS, <speechsc@ietf.org>
cc

Subject
RE: [Speechsc] Personal-Grammar-URI clarification






I am not sure what you mean by its relationship to Content-Id.

To me Personal-Grammar-URI is just like anyother grammar that has a
<one-of> list of items. 
Except that it is constructed by the user adding individual phrases to
that one-of list during enrollment.  Once enrolled, the usage of that
grammar should be no different from any other grammar.

This could be a HTTP URI in which case the MRCP server uses HTTP GET/PUT
to modify that grammar during enrollment. So to answer your question,
the MRCP server is not expected to store this personal grammar, but
would be create/modified at the URI specified by the personal grammar
URI.

Sarvi

     -----Original Message-----
     From: speechsc-bounces@ietf.org 
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
     Sent: Wednesday, August 10, 2005 11:43 AM
     To: speechsc@ietf.org
     Subject: Re: [Speechsc] Personal-Grammar-URI clarification
 
     Bump
 
     Thanks,
 
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
 
 
 
 
     Brett Gavagni/West Palm Beach/IBM@IBMUS Sent by: 
     speechsc-bounces@ietf.org
     07/26/2005 11:28 AM
 
     To
     speechsc@ietf.org
     cc
 
     Subject
     [Speechsc] Personal-Grammar-URI clarification
 
 
 
 
 
 
     Hi,
 
     The wording throughout the current draft specification is 
     a bit confusing w.r.t "Personal-Grammar-URI".
 
     Is the expectation that the "Personal-Grammar-URI" value 
     is merely a client specified handle to an external grammar 
     in lieu of using a "Content-Id" value?
 
     Is the expectation that the server persistently store 
     these references and 
 
     the backing grammars, and when and how would the release 
     of these references occur?
 
     Thanks,
 
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
 
 
     _______________________________________________
     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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From speechsc-bounces@ietf.org Wed Aug 17 12:06:48 2005
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Subject: RE: [Speechsc] Personal-Grammar-URI clarification
Date: Wed, 17 Aug 2005 08:52:44 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C3D57@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] Personal-Grammar-URI clarification
Thread-Index: AcWjQfxwQyRGkh52TD6qvphj/LPmJwAAZsYw
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Brett Gavagni" <gavagni@us.ibm.com>
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Don't understand. Could you please clarify and Propose some text.

Sarvi=20

     -----Original Message-----
     From: Brett Gavagni [mailto:gavagni@us.ibm.com]=20
     Sent: Wednesday, August 17, 2005 8:39 AM
     To: Shanmugham, Saravanan
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] Personal-Grammar-URI clarification
    =20
     Thanks for the clarification!
    =20
     Since SRGS defines syntactical support for both ABNF and=20
     XML formats, it may be useful to add some wording or=20
     modify the "Personal-Grammar-URI"=20
     examples to specify the "Content-Type".
    =20
     This may assist in facilitating client/server interoperability.=20
    =20
     Thanks,
    =20
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
    =20
    =20
    =20
    =20
     "Shanmugham, Saravanan" <sarvi@cisco.com>
     08/10/2005 04:03 PM
    =20
     To
     Brett Gavagni/West Palm Beach/IBM@IBMUS, <speechsc@ietf.org> cc
    =20
     Subject
     RE: [Speechsc] Personal-Grammar-URI clarification
    =20
    =20
    =20
    =20
    =20
    =20
     I am not sure what you mean by its relationship to Content-Id.
    =20
     To me Personal-Grammar-URI is just like anyother grammar=20
     that has a <one-of> list of items.=20
     Except that it is constructed by the user adding=20
     individual phrases to that one-of list during enrollment. =20
     Once enrolled, the usage of that grammar should be no=20
     different from any other grammar.
    =20
     This could be a HTTP URI in which case the MRCP server=20
     uses HTTP GET/PUT to modify that grammar during=20
     enrollment. So to answer your question, the MRCP server is=20
     not expected to store this personal grammar, but would be=20
     create/modified at the URI specified by the personal grammar URI.
    =20
     Sarvi
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org=20
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
          Sent: Wednesday, August 10, 2005 11:43 AM
          To: speechsc@ietf.org
          Subject: Re: [Speechsc] Personal-Grammar-URI clarification
     =20
          Bump
     =20
          Thanks,
     =20
          Brett Gavagni
          WebSphere Voice Server Development
          http://www-306.ibm.com/software/pervasive/voice_server/
          gavagni@us.ibm.com
     =20
     =20
     =20
     =20
          Brett Gavagni/West Palm Beach/IBM@IBMUS Sent by:=20
          speechsc-bounces@ietf.org
          07/26/2005 11:28 AM
     =20
          To
          speechsc@ietf.org
          cc
     =20
          Subject
          [Speechsc] Personal-Grammar-URI clarification
     =20
     =20
     =20
     =20
     =20
     =20
          Hi,
     =20
          The wording throughout the current draft specification is=20
          a bit confusing w.r.t "Personal-Grammar-URI".
     =20
          Is the expectation that the "Personal-Grammar-URI" value=20
          is merely a client specified handle to an external grammar=20
          in lieu of using a "Content-Id" value?
     =20
          Is the expectation that the server persistently store=20
          these references and=20
     =20
          the backing grammars, and when and how would the release=20
          of these references occur?
     =20
          Thanks,
     =20
          Brett Gavagni
          WebSphere Voice Server Development
          http://www-306.ibm.com/software/pervasive/voice_server/
          gavagni@us.ibm.com
     =20
     =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
     =20
     =20
     =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
     =20
    =20

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Subject: RE: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-07.txt 
Date: Wed, 17 Aug 2005 08:06:58 -0700
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Thread-Index: AcWW1lk3cgJ+fpmySA+LBlxcCdmm2gMYmAhgAAEXFBA=
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Pierre Forgues" <forgues@nuance.com>, <speechsc@ietf.org>
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That should text should go. My bad. Will take care of it in the next
rev.=20

But per dave's feed back this is pretty complex to maintain, we will not
be adding it back.

Sarvi=20

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Pierre Forgues
     Sent: Wednesday, August 17, 2005 7:36 AM
     To: speechsc@ietf.org
     Subject: RE: [Speechsc] I-D=20
     ACTION:draft-ietf-speechsc-mrcpv2-07.txt=20
    =20
     When is the MRCP draft document going to be revised with=20
     the synthesizer and recognizer usage tables included?  The=20
     following text in draft 07 needs to be replaced with the=20
     actual table (same for the recognizer
     resource):
    =20
        SYNTHESIZER HEADER USAGE TABLE TEMPORARILY REMOVED DUE=20
     TO XML2RFC
        LAYOUT PROBLEMS
    =20
     Thanks,
     Pierre
    =20
     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Internet-Drafts@ietf.org
     Sent: Monday, August 01, 2005 3:50 PM
     To: i-d-announce@ietf.org
     Cc: speechsc@ietf.org
     Subject: [Speechsc] I-D ACTION:draft-ietf-speechsc-mrcpv2-07.txt=20
    =20
     A New Internet-Draft is available from the on-line=20
     Internet-Drafts directories.
     This draft is a work item of the Speech Services Control=20
     Working Group of the IETF.
    =20
     	Title		: Media Resource Control Protocol Version 2
     (MRCPv2)
     	Author(s)	: D. Burnett, S. Shanmugham
     	Filename	: draft-ietf-speechsc-mrcpv2-07.txt
     	Pages		: 198
     	Date		: 2005-8-1
     =09
     The MRCPv2 protocol allows client hosts to control media service
        resources such as speech synthesizers, recognizers,=20
     verifiers and
        identifiers residing in servers on the network.  MRCPv2 is not a
        "stand-alone" protocol - it relies on a session=20
     management protocol
        such as the Session Initiation Protocol (SIP) to=20
     establish the MRCPv2
        control session between the client and the server, and=20
     for rendezvous
        and capability discovery.  It also depends on SIP and SDP to
        establish the media sessions and associated parameters=20
     between the
        media source or sink and the media server.  Once this=20
     is done, the
        MRCPv2 protocol exchange operates over the control session
        established above, allowing the client to control the media
        processing resources on the speech resource server.
    =20
     A URL for this Internet-Draft is:
     http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrc
     pv2-07.txt
    =20
     To remove yourself from the I-D Announcement list, send a=20
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    =20
    =20
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    =20
     A list of Internet-Drafts directories can be found in=20
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     ftp://ftp.ietf.org/ietf/1shadow-sites.txt
    =20
    =20
     Internet-Drafts can also be obtained by e-mail.
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     Send a message to:
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     =09
     NOTE:	The mail server at ietf.org can return the document in
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     	=09
     	=09
     Below is the data which will enable a MIME compliant mail reader
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    =20
     _______________________________________________
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     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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From speechsc-bounces@ietf.org Mon Aug 22 06:05:10 2005
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	"David R Oran" <oran@cisco.com>
References: <1424D740-50C9-4D7B-8AE5-40E7AF5E7681@cisco.com>
Subject: Re: [Speechsc] Driving toward Last Call on MRCPv2
Date: Mon, 22 Aug 2005 11:04:40 +0100
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Hello,

Version 7 looks good to me (nice job). Some bugs / features not being 
tracked as far as I can see:

Bugs:
-------

A. We decided to remove phrase "media server" - e.g. 
http://www.ietf.org/mail-archive/web/speechsc/current/msg00631.html. Looks 
like v7 has cleaned this up nicely except for the Abstract. Change "media 
service resources" to "media processing resources" and "media server" to 
"media processing server" for consistency with rest of text.

Features:
------------

B. Managing big grammars: 
http://www.ietf.org/mail-archive/web/speechsc/current/msg01404.html. What I 
believe was agreeable:
1. Note that MRCP servers may choose to centrally cache grammars
2. Note that the MRCP server may choose to use a stale grammar temporarily 
to avoid latency when a HTTP revalidation is compile but must return a 
Warning
3. Using Cache-Control: max-stale=0 overrides point 2
4. Grammar caching ought to be SHOULD not MAY
5. Are DEFINE-GRAMMAR handles (Content-Id) global? On one hand, we don't 
want to have to ship large payloads to the MRCP server every call (e.g. if 
grammar is inline). On the other, does an MRCP server have to remember the 
Content-Id<->URI or Content-Id<-> document mapping forever? So perhaps it 
has to be the former. At least the operation should be clarified...

C. START-OF-INPUT to include a InputType field (dtmf or speech) - 
http://www1.ietf.org/mail-archive/web/speechsc/current/msg01454.html

D. Track this at least for a future version. There is no mechanism for 
linking a control channel to more than one media stream as would be 
necessary for a video evolution. One simple approach (which at least 
illustrates the point) is to have a a=cmid:1,2 where a=mid:1 is an audio 
stream and a=mid:2 is a video stream. Alternatively use RFC 3388. We have a 
need for this today for support video media in VoiceXML. See - 
http://www.ietf.org/mail-archive/web/speechsc/current/msg01281.html

- Dave


----- Original Message ----- 
From: "David R Oran" <oran@cisco.com>
To: "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
Sent: Monday, August 08, 2005 7:34 PM
Subject: [Speechsc] Driving toward Last Call on MRCPv2


> By now you folks should have noticed that -07 was posted by Sarvi.  This 
> was a major revision, incorporating resolutions to the vast  majority of 
> the open issues, a conversion from Word to XML source,  and a 
> comprehensive technical and editorial review by both co-chairs.
>
> If you haven't already, you can get it from:
> http://www.ietf.org/internet-drafts/draft-ietf-speechsc-mrcpv2-07.txt
>
> In order to keep track of the issues raised on the mailing list and  to 
> manage the editing process, the chairs and authors have been using  the 
> "roundup" issue tracking tool. It is being hosted on  softarmor.com with 
> the gracious help of Dean Willis. We kept it  private for the creation 
> of -07, but are now ready to open it to the  whole working group to review 
> the changes and see the status of the  remaining open issues.
>
> Please navigate to https://www.softarmor.com/roundup/speechsc/ and 
> register yourself to access the issues list.
>
> The chairs would like to have no more than one further revision of  the 
> specification and be ready for WG last call by the end of August. 
> Therefore, please:
> - make sure any issues you have reaised and were supposed to be  addressed 
> in -07 in fact have been marked resolved in the issue  tracker and 
> correctly reflected in -07
> - review the open issues and indicate on the mailing list if there  you 
> feel their are any which MUST be resolved prior to going to last  call. 
> Unless we hear otherwise, we would like to defer non-critical 
> functionality from this version of MRCP so we can get to proposed 
> standard soon.
>
> We would like to declare rough consensus on each of the open issues  as 
> soon as feasible, and surface any new issues and deal with them 
> expeditiously.
>
> I know this is mid-summer and people have lots of other things to  think 
> about. Some of your time and energy on this effort will be of  great 
> assistance to the working group in completing our main piece of  technical 
> work with minimal further delay.
>
> Thanks and Regards,
> Dave Oran (speechsc co-chair).
>
>
> _______________________________________________
> Speechsc mailing list
> Speechsc@ietf.org
> https://www1.ietf.org/mailman/listinfo/speechsc
> 


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Subject: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
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Hi,

What's the status of this request that was made a few months ago, as it 
doesn't appear to have been addressed in the -07 draft?

Here's the thread on the topic:
http://www1.ietf.org/mail-archive/web/speechsc/current/msg01234.html

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/
gavagni@us.ibm.com


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To: "Shanmugham, Saravanan" <sarvi@cisco.com>
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Subject: RE: [Speechsc] Personal-Grammar-URI clarification
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Thanks!

I see the point, the MRCP server's  HTTP PUT request should specify the 
"Content-Type" of "application/srgs" or "application/srgs+xml" as 
described in SRGS to indicate the grammar format.

At the time, I was suggesting to add some wording similar to the following 
to section 9.4.35  Personal-Grammar-URI:
The generated SRGS grammar syntax MAY be implementation specific.

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/
(561) 862-2097 T/L (975) 
gavagni@us.ibm.com




"Shanmugham, Saravanan" <sarvi@cisco.com> 
08/17/2005 11:52 AM

To
Brett Gavagni/West Palm Beach/IBM@IBMUS
cc
<speechsc@ietf.org>
Subject
RE: [Speechsc] Personal-Grammar-URI clarification






Don't understand. Could you please clarify and Propose some text.

Sarvi 

     -----Original Message-----
     From: Brett Gavagni [mailto:gavagni@us.ibm.com] 
     Sent: Wednesday, August 17, 2005 8:39 AM
     To: Shanmugham, Saravanan
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] Personal-Grammar-URI clarification
 
     Thanks for the clarification!
 
     Since SRGS defines syntactical support for both ABNF and 
     XML formats, it may be useful to add some wording or 
     modify the "Personal-Grammar-URI" 
     examples to specify the "Content-Type".
 
     This may assist in facilitating client/server interoperability. 
 
     Thanks,
 
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
 
 
 
 
     "Shanmugham, Saravanan" <sarvi@cisco.com>
     08/10/2005 04:03 PM
 
     To
     Brett Gavagni/West Palm Beach/IBM@IBMUS, <speechsc@ietf.org> cc
 
     Subject
     RE: [Speechsc] Personal-Grammar-URI clarification
 
 
 
 
 
 
     I am not sure what you mean by its relationship to Content-Id.
 
     To me Personal-Grammar-URI is just like anyother grammar 
     that has a <one-of> list of items. 
     Except that it is constructed by the user adding 
     individual phrases to that one-of list during enrollment. 
     Once enrolled, the usage of that grammar should be no 
     different from any other grammar.
 
     This could be a HTTP URI in which case the MRCP server 
     uses HTTP GET/PUT to modify that grammar during 
     enrollment. So to answer your question, the MRCP server is 
     not expected to store this personal grammar, but would be 
     create/modified at the URI specified by the personal grammar URI.
 
     Sarvi
 
          -----Original Message-----
          From: speechsc-bounces@ietf.org 
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
          Sent: Wednesday, August 10, 2005 11:43 AM
          To: speechsc@ietf.org
          Subject: Re: [Speechsc] Personal-Grammar-URI clarification
 
          Bump
 
          Thanks,
 
          Brett Gavagni
          WebSphere Voice Server Development
          http://www-306.ibm.com/software/pervasive/voice_server/
          gavagni@us.ibm.com
 
 
 
 
          Brett Gavagni/West Palm Beach/IBM@IBMUS Sent by: 
          speechsc-bounces@ietf.org
          07/26/2005 11:28 AM
 
          To
          speechsc@ietf.org
          cc
 
          Subject
          [Speechsc] Personal-Grammar-URI clarification
 
 
 
 
 
 
          Hi,
 
          The wording throughout the current draft specification is 
          a bit confusing w.r.t "Personal-Grammar-URI".
 
          Is the expectation that the "Personal-Grammar-URI" value 
          is merely a client specified handle to an external grammar 
          in lieu of using a "Content-Id" value?
 
          Is the expectation that the server persistently store 
          these references and 
 
          the backing grammars, and when and how would the release 
          of these references occur?
 
          Thanks,
 
          Brett Gavagni
          WebSphere Voice Server Development
          http://www-306.ibm.com/software/pervasive/voice_server/
          gavagni@us.ibm.com
 
 
          _______________________________________________
          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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From speechsc-bounces@ietf.org Mon Aug 22 13:25:33 2005
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Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Date: Mon, 22 Aug 2005 10:25:17 -0700
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From: "Shanmugham, Saravanan" <sarvi@cisco.com>
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Status in Issue Tracker.
---------------------------
 Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
Added an Input-Type header to all input processing resources, with
values
"speech" and "dtmf"
Add text to explain that what constitutes a barge-in event based on the
resource
type and the grammars that are active.  START-OF-INPUT event is
generated with
this barge-in condition is met.


Sarvi



     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
     Sent: Monday, August 22, 2005 7:53 AM
     To: speechsc@ietf.org
     Subject: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes" property
    =20
     Hi,
    =20
     What's the status of this request that was made a few=20
     months ago, as it doesn't appear to have been addressed in=20
     the -07 draft?
    =20
     Here's the thread on the topic:
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
     1234.html
    =20
     Thanks,
    =20
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

_______________________________________________
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In-Reply-To: <03772D1EC8DE624A863058C75874A75C2C4046@vtg-um-e2k6.sj21ad.cisco.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>
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It seems a bit inefficient for a VoiceXML platform utilizing MRCP to rely 
on the START-OF-INPUT to determine if the identified "Input-Type" is 
enabled in the current VoiceXML application execution context. 

I would suggest that a MRCP client be able to specify the input mode 
during the RECOGNIZE request.

Additionally, including the mode that MRCP server's grammar processor 
interpreted for each DEFINE-GRAMMAR request would assist the MRCP client 
with enabling the ideal set of grammars.

Thanks,

Brett Gavagni 
WebSphere Voice Server Development 
http://www-306.ibm.com/software/pervasive/voice_server/
gavagni@us.ibm.com




"Shanmugham, Saravanan" <sarvi@cisco.com> 
08/22/2005 01:25 PM

To
Brett Gavagni/West Palm Beach/IBM@IBMUS, <speechsc@ietf.org>
cc

Subject
RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property






Status in Issue Tracker.
---------------------------
 Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
Added an Input-Type header to all input processing resources, with
values
"speech" and "dtmf"
Add text to explain that what constitutes a barge-in event based on the
resource
type and the grammars that are active.  START-OF-INPUT event is
generated with
this barge-in condition is met.


Sarvi



     -----Original Message-----
     From: speechsc-bounces@ietf.org 
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
     Sent: Monday, August 22, 2005 7:53 AM
     To: speechsc@ietf.org
     Subject: [Speechsc] MRCPv2 requirements for VoiceXML 
     "inputmodes" property
 
     Hi,
 
     What's the status of this request that was made a few 
     months ago, as it doesn't appear to have been addressed in 
     the -07 draft?
 
     Here's the thread on the topic:
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
     1234.html
 
     Thanks,
 
     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com
 
 
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
 



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From speechsc-bounces@ietf.org Tue Aug 23 01:45:48 2005
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From: "Dave Burke" <david.burke@voxpilot.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
References: <03772D1EC8DE624A863058C75874A75C2C4046@vtg-um-e2k6.sj21ad.cisco.com>
Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Date: Tue, 23 Aug 2005 06:45:22 +0100
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Sarvi,

The changes below look mostly fine to me with the bug noted (point C in
http://www1.ietf.org/mail-archive/web/speechsc/current/msg01485.html).

One concern I do have with it is that -07 says about the speechrecog 
resource:

"This resource generates barge-in events for
   speech.  By analyzing the grammars that are activated by the
   RECOGNIZE method, it also figures out if a barge-in should occur for
   DTMF or not."

whereas I thought it ought to say (based on our discussions):

   "This resource generates barge-in events for
   speech and/or DTMF.  By analyzing the grammar types that are activated
   by the RECOGNIZE method, it determines  if a barge-in should occur for
   speech and/or DTMF"

The concern I have with the current wording is that during a call, it is
common to go from speech+dtmf reco to dtmf-only reco and then back to
speech+dtmf. With current wording, one would need to perform a REINVITE
to change from a speechrecog to a dtmfrecog. To revert to speech+dtmf,
another REINVITEs is needed. So 2 REINVITEs (=6 SIP messages) would be
needed to handle the common use-case for DTMF-only fallback.

Dave

----- Original Message ----- 
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Brett Gavagni" <gavagni@us.ibm.com>; <speechsc@ietf.org>
Sent: Monday, August 22, 2005 6:25 PM
Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" 
property


Status in Issue Tracker.
---------------------------
 Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
Added an Input-Type header to all input processing resources, with
values
"speech" and "dtmf"
Add text to explain that what constitutes a barge-in event based on the
resource
type and the grammars that are active.  START-OF-INPUT event is
generated with
this barge-in condition is met.


Sarvi



     -----Original Message-----
     From: speechsc-bounces@ietf.org
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
     Sent: Monday, August 22, 2005 7:53 AM
     To: speechsc@ietf.org
     Subject: [Speechsc] MRCPv2 requirements for VoiceXML
     "inputmodes" property

     Hi,

     What's the status of this request that was made a few
     months ago, as it doesn't appear to have been addressed in
     the -07 draft?

     Here's the thread on the topic:
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
     1234.html

     Thanks,

     Brett Gavagni
     WebSphere Voice Server Development
     http://www-306.ibm.com/software/pervasive/voice_server/
     gavagni@us.ibm.com


     _______________________________________________
     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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From speechsc-bounces@ietf.org Tue Aug 23 02:20:18 2005
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Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Date: Mon, 22 Aug 2005 23:17:17 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C4127@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Thread-Index: AcWnpjwzgCzYZoZlSp+QcIN1yoCyWwAA/1Jg
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Dave Burke" <david.burke@voxpilot.com>,
	"Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
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The difference, I see between the 2 is or Vs and/or. I can certainly
make that change.
But I don't understand your SIP INVITE deal.

This choice of dtmf and/or speech for barge-input is based on a per
RECOGNIZE method.=20
That is where we activate recognition and specify a list of grammars to
activate. I was thinking that's where this choice would be made based on
the list of grammars being activated.=20

I don't see any SIP re-INVITES happenning in this case. Am I missing
something.

Sarvi=20
=20

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Dave Burke
     Sent: Monday, August 22, 2005 10:45 PM
     To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
     Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes" property
    =20
     Sarvi,
    =20
     The changes below look mostly fine to me with the bug=20
     noted (point C in=20
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
     1485.html).
    =20
     One concern I do have with it is that -07 says about the=20
     speechrecog
     resource:
    =20
     "This resource generates barge-in events for
        speech.  By analyzing the grammars that are activated by the
        RECOGNIZE method, it also figures out if a barge-in=20
     should occur for
        DTMF or not."
    =20
     whereas I thought it ought to say (based on our discussions):
    =20
        "This resource generates barge-in events for
        speech and/or DTMF.  By analyzing the grammar types=20
     that are activated
        by the RECOGNIZE method, it determines  if a barge-in=20
     should occur for
        speech and/or DTMF"
    =20
     The concern I have with the current wording is that during=20
     a call, it is common to go from speech+dtmf reco to=20
     dtmf-only reco and then back to
     speech+dtmf. With current wording, one would need to=20
     perform a REINVITE
     to change from a speechrecog to a dtmfrecog. To revert to=20
     speech+dtmf, another REINVITEs is needed. So 2 REINVITEs=20
     (=3D6 SIP messages) would be needed to handle the common=20
     use-case for DTMF-only fallback.
    =20
     Dave
    =20
     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Brett Gavagni" <gavagni@us.ibm.com>; <speechsc@ietf.org>
     Sent: Monday, August 22, 2005 6:25 PM
     Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes"=20
     property
    =20
    =20
     Status in Issue Tracker.
     ---------------------------
      Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
     Added an Input-Type header to all input processing resources, with
     values
     "speech" and "dtmf"
     Add text to explain that what constitutes a barge-in event=20
     based on the
     resource
     type and the grammars that are active.  START-OF-INPUT event is
     generated with
     this barge-in condition is met.
    =20
    =20
     Sarvi
    =20
    =20
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
          Sent: Monday, August 22, 2005 7:53 AM
          To: speechsc@ietf.org
          Subject: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes" property
    =20
          Hi,
    =20
          What's the status of this request that was made a few
          months ago, as it doesn't appear to have been addressed in
          the -07 draft?
    =20
          Here's the thread on the topic:
          http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
          1234.html
    =20
          Thanks,
    =20
          Brett Gavagni
          WebSphere Voice Server Development
          http://www-306.ibm.com/software/pervasive/voice_server/
          gavagni@us.ibm.com
    =20
    =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

_______________________________________________
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From speechsc-bounces@ietf.org Tue Aug 23 03:02:16 2005
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From: "Dave Burke" <david.burke@voxpilot.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
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Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Date: Tue, 23 Aug 2005 08:01:57 +0100
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In the -07 version, the wording implies that the speechrecog will always 
barge-in on speech input regardless of whether a speech grammar was 
activated in the RECOGNIZE.

So imagine a client has connected to a speechrecog resource. At a certain 
point, it wants to perform a DTMF-only recognition (because it's noisy). If 
it performs a RECOGNIZE and activates just a DTMF grammar,  the -07 implies 
that the speechrecog will still barge-in on speech input (in this case the 
noise). The only way to perform a DTMF-only recognition with the -07 version 
is to change the resource type from a speechrecog to a dtmfrecog by 
performing a re-INVITE (altering the SDP by changing the 
a=resource:speechrecog to a=resource:dtmfrecog).

If we change the wording as suggested below, then it will work with needing 
a re-INVITE.

Dave

----- Original Message ----- 
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Dave Burke" <david.burke@voxpilot.com>; "Brett Gavagni" 
<gavagni@us.ibm.com>; <speechsc@ietf.org>
Sent: Tuesday, August 23, 2005 7:17 AM
Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" 
property


The difference, I see between the 2 is or Vs and/or. I can certainly
make that change.
But I don't understand your SIP INVITE deal.

This choice of dtmf and/or speech for barge-input is based on a per
RECOGNIZE method.
That is where we activate recognition and specify a list of grammars to
activate. I was thinking that's where this choice would be made based on
the list of grammars being activated.

I don't see any SIP re-INVITES happenning in this case. Am I missing
something.

Sarvi


     -----Original Message-----
     From: speechsc-bounces@ietf.org
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Dave Burke
     Sent: Monday, August 22, 2005 10:45 PM
     To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
     Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML
     "inputmodes" property

     Sarvi,

     The changes below look mostly fine to me with the bug
     noted (point C in
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
     1485.html).

     One concern I do have with it is that -07 says about the
     speechrecog
     resource:

     "This resource generates barge-in events for
        speech.  By analyzing the grammars that are activated by the
        RECOGNIZE method, it also figures out if a barge-in
     should occur for
        DTMF or not."

     whereas I thought it ought to say (based on our discussions):

        "This resource generates barge-in events for
        speech and/or DTMF.  By analyzing the grammar types
     that are activated
        by the RECOGNIZE method, it determines  if a barge-in
     should occur for
        speech and/or DTMF"

     The concern I have with the current wording is that during
     a call, it is common to go from speech+dtmf reco to
     dtmf-only reco and then back to
     speech+dtmf. With current wording, one would need to
     perform a REINVITE
     to change from a speechrecog to a dtmfrecog. To revert to
     speech+dtmf, another REINVITEs is needed. So 2 REINVITEs
     (=6 SIP messages) would be needed to handle the common
     use-case for DTMF-only fallback.

     Dave

     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Brett Gavagni" <gavagni@us.ibm.com>; <speechsc@ietf.org>
     Sent: Monday, August 22, 2005 6:25 PM
     Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML
     "inputmodes"
     property


     Status in Issue Tracker.
     ---------------------------
      Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
     Added an Input-Type header to all input processing resources, with
     values
     "speech" and "dtmf"
     Add text to explain that what constitutes a barge-in event
     based on the
     resource
     type and the grammars that are active.  START-OF-INPUT event is
     generated with
     this barge-in condition is met.


     Sarvi



          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Brett Gavagni
          Sent: Monday, August 22, 2005 7:53 AM
          To: speechsc@ietf.org
          Subject: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes" property

          Hi,

          What's the status of this request that was made a few
          months ago, as it doesn't appear to have been addressed in
          the -07 draft?

          Here's the thread on the topic:
          http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
          1234.html

          Thanks,

          Brett Gavagni
          WebSphere Voice Server Development
          http://www-306.ibm.com/software/pervasive/voice_server/
          gavagni@us.ibm.com


          _______________________________________________
          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


     _______________________________________________
     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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From: Andrew Wahbe <awahbe@voicegenie.com>
Organization: VoiceGenie Technologies
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Subject: [Speechsc] Recognition Completion-Cause Codes 003 and 008 revisited
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A few weeks ago, there was an extensive discussion about the recognition 
completion-cause codes for cases where recognition-timers fire. There 
seemed to be two codes, 003 and 008, that meant the same thing -- the 
recognition was aborted due to the recognition-timer firing. This case 
maps to the maxspeechtimeout event in VoiceXML.

A lot of hypothesizing was done to try and figure out what the 
difference between the codes was. After some discussion it seems we 
ended up with no codes that map to maxspeechtimeout in VoiceXML (i.e. 
the recognition was aborted because of recognition-timeout and there is 
no match).

I think things would be set right if 003 was changed from:

   | 003    | recognition-timeout   | RECOGNIZE in hotword mode        |
   |        |                       | completed without a match due to |
   |        |                       | a recognition-timeout            |

to:

   | 003    | recognition-timeout   | RECOGNIZE completed without a    |
   |        |                       | match due to a                   |
   |        |                       | recognition-timeout              |


I am not sure why a special code is needed for this timer firing in 
hotword mode. Am I missing something?

Andrew Wahbe
VoiceGenie Technologies

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From speechsc-bounces@ietf.org Tue Aug 23 11:21:13 2005
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Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Date: Tue, 23 Aug 2005 08:18:27 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C413C@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Thread-Index: AcWnsJl0uTe7tL3oRSyAMbS/F7E5rAARLBBg
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Dave Burke" <david.burke@voxpilot.com>,
	"Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
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The speech syntheisizer barges-in when the it receives the
BARGEIN-OCCURRED method from the client, or if the recognizer on the
same session generated a START-OF-INPUTE event, in the optimized
barge-in case.

The question though, is when does the START-OF-INPUT event get
generated. And based on the current wording , the recognizer should be
smart enough to generate only one START-OF-INPUT event based on the
grammars that are active for that RECOGNIZE method.

Sarvi=20

     -----Original Message-----
     From: Dave Burke [mailto:david.burke@voxpilot.com]=20
     Sent: Tuesday, August 23, 2005 12:02 AM
     To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
     Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes" property
    =20
     In the -07 version, the wording implies that the=20
     speechrecog will always barge-in on speech input=20
     regardless of whether a speech grammar was activated in=20
     the RECOGNIZE.
    =20
     So imagine a client has connected to a speechrecog=20
     resource. At a certain point, it wants to perform a=20
     DTMF-only recognition (because it's noisy). If it performs=20
     a RECOGNIZE and activates just a DTMF grammar,  the -07=20
     implies that the speechrecog will still barge-in on speech=20
     input (in this case the noise). The only way to perform a=20
     DTMF-only recognition with the -07 version is to change=20
     the resource type from a speechrecog to a dtmfrecog by=20
     performing a re-INVITE (altering the SDP by changing the=20
     a=3Dresource:speechrecog to a=3Dresource:dtmfrecog).
    =20
     If we change the wording as suggested below, then it will=20
     work with needing a re-INVITE.
    =20
     Dave
    =20
     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Dave Burke" <david.burke@voxpilot.com>; "Brett Gavagni"=20
     <gavagni@us.ibm.com>; <speechsc@ietf.org>
     Sent: Tuesday, August 23, 2005 7:17 AM
     Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes"=20
     property
    =20
    =20
     The difference, I see between the 2 is or Vs and/or. I can=20
     certainly
     make that change.
     But I don't understand your SIP INVITE deal.
    =20
     This choice of dtmf and/or speech for barge-input is based on a per
     RECOGNIZE method.
     That is where we activate recognition and specify a list=20
     of grammars to
     activate. I was thinking that's where this choice would be=20
     made based on
     the list of grammars being activated.
    =20
     I don't see any SIP re-INVITES happenning in this case. Am=20
     I missing
     something.
    =20
     Sarvi
    =20
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Dave Burke
          Sent: Monday, August 22, 2005 10:45 PM
          To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
          Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes" property
    =20
          Sarvi,
    =20
          The changes below look mostly fine to me with the bug
          noted (point C in
          http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
          1485.html).
    =20
          One concern I do have with it is that -07 says about the
          speechrecog
          resource:
    =20
          "This resource generates barge-in events for
             speech.  By analyzing the grammars that are=20
     activated by the
             RECOGNIZE method, it also figures out if a barge-in
          should occur for
             DTMF or not."
    =20
          whereas I thought it ought to say (based on our discussions):
    =20
             "This resource generates barge-in events for
             speech and/or DTMF.  By analyzing the grammar types
          that are activated
             by the RECOGNIZE method, it determines  if a barge-in
          should occur for
             speech and/or DTMF"
    =20
          The concern I have with the current wording is that during
          a call, it is common to go from speech+dtmf reco to
          dtmf-only reco and then back to
          speech+dtmf. With current wording, one would need to
          perform a REINVITE
          to change from a speechrecog to a dtmfrecog. To revert to
          speech+dtmf, another REINVITEs is needed. So 2 REINVITEs
          (=3D6 SIP messages) would be needed to handle the common
          use-case for DTMF-only fallback.
    =20
          Dave
    =20
          ----- Original Message -----
          From: "Shanmugham, Saravanan" <sarvi@cisco.com>
          To: "Brett Gavagni" <gavagni@us.ibm.com>; <speechsc@ietf.org>
          Sent: Monday, August 22, 2005 6:25 PM
          Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes"
          property
    =20
    =20
          Status in Issue Tracker.
          ---------------------------
           Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
          Added an Input-Type header to all input processing=20
     resources, with
          values
          "speech" and "dtmf"
          Add text to explain that what constitutes a barge-in event
          based on the
          resource
          type and the grammars that are active. =20
     START-OF-INPUT event is
          generated with
          this barge-in condition is met.
    =20
    =20
          Sarvi
    =20
    =20
    =20
               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Brett Gavagni
               Sent: Monday, August 22, 2005 7:53 AM
               To: speechsc@ietf.org
               Subject: [Speechsc] MRCPv2 requirements for VoiceXML
               "inputmodes" property
    =20
               Hi,
    =20
               What's the status of this request that was made a few
               months ago, as it doesn't appear to have been=20
     addressed in
               the -07 draft?
    =20
               Here's the thread on the topic:
              =20
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
               1234.html
    =20
               Thanks,
    =20
               Brett Gavagni
               WebSphere Voice Server Development
               http://www-306.ibm.com/software/pervasive/voice_server/
               gavagni@us.ibm.com
    =20
    =20
               _______________________________________________
               Speechsc mailing list
               Speechsc@ietf.org
               https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
          _______________________________________________
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          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
          _______________________________________________
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          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
     _______________________________________________
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     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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To: "'sarvi@cisco.com'" <sarvi@cisco.com>, "'dburnett@vocalocity.net'"
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Subject: [Speechsc] Problem with DTMF type-ahead 
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The MRCP and MRCPv2 specifications define how DTMF input should be handled:
DTMF input is collected after the RECOGNIZE request is received until a
term-char or timeout is reached. Then a result or no-match is returned based
on any active DTMF grammars. The problem is there can be a significant
amount of time between when one result is processed by the application and
the next grammar is activated. During that time any DTMF input received will
be lost. Such DTMF type-ahead is common for frequent users of IVR systems,
and buffering that type ahead is required by VoiceXML 2.0 (see section
4.1.8). Can this requirement of VoiceXML be implemented using MRCP (assuming
that the media source CANNOT buffer DTMF)? If so, how is it done? 
Thanks,
Klaus


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<P><FONT SIZE=3D2 FACE=3D"Arial">The MRCP and MRCPv2 specifications =
define how DTMF input should be handled: DTMF input is collected after =
the RECOGNIZE request is received until a term-char or timeout is =
reached. Then a result or no-match is returned based on any active DTMF =
grammars. The problem is there can be a significant amount of time =
between when one result is processed by the application and the next =
grammar is activated. During that time any DTMF input received will be =
lost. Such DTMF type-ahead is common for frequent users of IVR systems, =
and buffering that type ahead is required by VoiceXML 2.0 (see section =
4.1.8).</FONT><FONT FACE=3D"Times New Roman"> Can</FONT><FONT SIZE=3D2 =
FACE=3D"Arial"> this requirement of VoiceXML be implemented using MRCP =
(assuming that the media source CANNOT buffer DTMF)? If so, how is it =
done? </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thanks,<BR>
Klaus</FONT>
</P>

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From: "Dave Burke" <david.burke@voxpilot.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
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Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
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Right - I was talking about the optimised case. Looks like we both agree on 
the mechanism. My problem is only with the wording "This resource generates 
barge-in events for speech" and "it also figures out if a barge-in should 
occur for DTMF".

This reads to me that the speechrecog *always* generates a START-OF-INPUT 
for any speech/noise heard regardless of what grammar types were activated 
in the RECOGNIZE. The small text change suggested below removes the 
ambiguity. Hope I'm being clear.

Thanks,

Dave

----- Original Message ----- 
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Dave Burke" <david.burke@voxpilot.com>; "Brett Gavagni" 
<gavagni@us.ibm.com>; <speechsc@ietf.org>
Sent: Tuesday, August 23, 2005 4:18 PM
Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" 
property


The speech syntheisizer barges-in when the it receives the
BARGEIN-OCCURRED method from the client, or if the recognizer on the
same session generated a START-OF-INPUTE event, in the optimized
barge-in case.

The question though, is when does the START-OF-INPUT event get
generated. And based on the current wording , the recognizer should be
smart enough to generate only one START-OF-INPUT event based on the
grammars that are active for that RECOGNIZE method.

Sarvi

     -----Original Message-----
     From: Dave Burke [mailto:david.burke@voxpilot.com]
     Sent: Tuesday, August 23, 2005 12:02 AM
     To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
     Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML
     "inputmodes" property

     In the -07 version, the wording implies that the
     speechrecog will always barge-in on speech input
     regardless of whether a speech grammar was activated in
     the RECOGNIZE.

     So imagine a client has connected to a speechrecog
     resource. At a certain point, it wants to perform a
     DTMF-only recognition (because it's noisy). If it performs
     a RECOGNIZE and activates just a DTMF grammar,  the -07
     implies that the speechrecog will still barge-in on speech
     input (in this case the noise). The only way to perform a
     DTMF-only recognition with the -07 version is to change
     the resource type from a speechrecog to a dtmfrecog by
     performing a re-INVITE (altering the SDP by changing the
     a=resource:speechrecog to a=resource:dtmfrecog).

     If we change the wording as suggested below, then it will
     work with needing a re-INVITE.

     Dave

     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Dave Burke" <david.burke@voxpilot.com>; "Brett Gavagni"
     <gavagni@us.ibm.com>; <speechsc@ietf.org>
     Sent: Tuesday, August 23, 2005 7:17 AM
     Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML
     "inputmodes"
     property


     The difference, I see between the 2 is or Vs and/or. I can
     certainly
     make that change.
     But I don't understand your SIP INVITE deal.

     This choice of dtmf and/or speech for barge-input is based on a per
     RECOGNIZE method.
     That is where we activate recognition and specify a list
     of grammars to
     activate. I was thinking that's where this choice would be
     made based on
     the list of grammars being activated.

     I don't see any SIP re-INVITES happenning in this case. Am
     I missing
     something.

     Sarvi


          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Dave Burke
          Sent: Monday, August 22, 2005 10:45 PM
          To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
          Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes" property

          Sarvi,

          The changes below look mostly fine to me with the bug
          noted (point C in
          http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
          1485.html).

          One concern I do have with it is that -07 says about the
          speechrecog
          resource:

          "This resource generates barge-in events for
             speech.  By analyzing the grammars that are
     activated by the
             RECOGNIZE method, it also figures out if a barge-in
          should occur for
             DTMF or not."

          whereas I thought it ought to say (based on our discussions):

             "This resource generates barge-in events for
             speech and/or DTMF.  By analyzing the grammar types
          that are activated
             by the RECOGNIZE method, it determines  if a barge-in
          should occur for
             speech and/or DTMF"

          The concern I have with the current wording is that during
          a call, it is common to go from speech+dtmf reco to
          dtmf-only reco and then back to
          speech+dtmf. With current wording, one would need to
          perform a REINVITE
          to change from a speechrecog to a dtmfrecog. To revert to
          speech+dtmf, another REINVITEs is needed. So 2 REINVITEs
          (=6 SIP messages) would be needed to handle the common
          use-case for DTMF-only fallback.

          Dave

          ----- Original Message -----
          From: "Shanmugham, Saravanan" <sarvi@cisco.com>
          To: "Brett Gavagni" <gavagni@us.ibm.com>; <speechsc@ietf.org>
          Sent: Monday, August 22, 2005 6:25 PM
          Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes"
          property


          Status in Issue Tracker.
          ---------------------------
           Addressed this by changing START-OF-SPEECH to START-OF-INPUT.
          Added an Input-Type header to all input processing
     resources, with
          values
          "speech" and "dtmf"
          Add text to explain that what constitutes a barge-in event
          based on the
          resource
          type and the grammars that are active.
     START-OF-INPUT event is
          generated with
          this barge-in condition is met.


          Sarvi



               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of
     Brett Gavagni
               Sent: Monday, August 22, 2005 7:53 AM
               To: speechsc@ietf.org
               Subject: [Speechsc] MRCPv2 requirements for VoiceXML
               "inputmodes" property

               Hi,

               What's the status of this request that was made a few
               months ago, as it doesn't appear to have been
     addressed in
               the -07 draft?

               Here's the thread on the topic:

     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
               1234.html

               Thanks,

               Brett Gavagni
               WebSphere Voice Server Development
               http://www-306.ibm.com/software/pervasive/voice_server/
               gavagni@us.ibm.com


               _______________________________________________
               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


          _______________________________________________
          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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From speechsc-bounces@ietf.org Tue Aug 23 12:39:30 2005
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Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Date: Tue, 23 Aug 2005 09:39:14 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C4150@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] MRCPv2 requirements for VoiceXML "inputmodes" property
Thread-Index: AcWn/sUzQH2+HqPdQXSpU/dtXLpcXQAAlKiw
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Dave Burke" <david.burke@voxpilot.com>,
	"Brett Gavagni" <gavagni@us.ibm.com>, <speechsc@ietf.org>
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If we agree on the concept I can clarify the text.

Sarvi=20

     -----Original Message-----
     From: Dave Burke [mailto:david.burke@voxpilot.com]=20
     Sent: Tuesday, August 23, 2005 9:20 AM
     To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
     Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes" property
    =20
     Right - I was talking about the optimised case. Looks like=20
     we both agree on the mechanism. My problem is only with=20
     the wording "This resource generates barge-in events for=20
     speech" and "it also figures out if a barge-in should=20
     occur for DTMF".
    =20
     This reads to me that the speechrecog *always* generates a=20
     START-OF-INPUT for any speech/noise heard regardless of=20
     what grammar types were activated in the RECOGNIZE. The=20
     small text change suggested below removes the ambiguity.=20
     Hope I'm being clear.
    =20
     Thanks,
    =20
     Dave
    =20
     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Dave Burke" <david.burke@voxpilot.com>; "Brett Gavagni"=20
     <gavagni@us.ibm.com>; <speechsc@ietf.org>
     Sent: Tuesday, August 23, 2005 4:18 PM
     Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML=20
     "inputmodes"=20
     property
    =20
    =20
     The speech syntheisizer barges-in when the it receives the
     BARGEIN-OCCURRED method from the client, or if the=20
     recognizer on the
     same session generated a START-OF-INPUTE event, in the optimized
     barge-in case.
    =20
     The question though, is when does the START-OF-INPUT event get
     generated. And based on the current wording , the=20
     recognizer should be
     smart enough to generate only one START-OF-INPUT event based on the
     grammars that are active for that RECOGNIZE method.
    =20
     Sarvi
    =20
          -----Original Message-----
          From: Dave Burke [mailto:david.burke@voxpilot.com]
          Sent: Tuesday, August 23, 2005 12:02 AM
          To: Shanmugham, Saravanan; Brett Gavagni; speechsc@ietf.org
          Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes" property
    =20
          In the -07 version, the wording implies that the
          speechrecog will always barge-in on speech input
          regardless of whether a speech grammar was activated in
          the RECOGNIZE.
    =20
          So imagine a client has connected to a speechrecog
          resource. At a certain point, it wants to perform a
          DTMF-only recognition (because it's noisy). If it performs
          a RECOGNIZE and activates just a DTMF grammar,  the -07
          implies that the speechrecog will still barge-in on speech
          input (in this case the noise). The only way to perform a
          DTMF-only recognition with the -07 version is to change
          the resource type from a speechrecog to a dtmfrecog by
          performing a re-INVITE (altering the SDP by changing the
          a=3Dresource:speechrecog to a=3Dresource:dtmfrecog).
    =20
          If we change the wording as suggested below, then it will
          work with needing a re-INVITE.
    =20
          Dave
    =20
          ----- Original Message -----
          From: "Shanmugham, Saravanan" <sarvi@cisco.com>
          To: "Dave Burke" <david.burke@voxpilot.com>; "Brett Gavagni"
          <gavagni@us.ibm.com>; <speechsc@ietf.org>
          Sent: Tuesday, August 23, 2005 7:17 AM
          Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML
          "inputmodes"
          property
    =20
    =20
          The difference, I see between the 2 is or Vs and/or. I can
          certainly
          make that change.
          But I don't understand your SIP INVITE deal.
    =20
          This choice of dtmf and/or speech for barge-input is=20
     based on a per
          RECOGNIZE method.
          That is where we activate recognition and specify a list
          of grammars to
          activate. I was thinking that's where this choice would be
          made based on
          the list of grammars being activated.
    =20
          I don't see any SIP re-INVITES happenning in this case. Am
          I missing
          something.
    =20
          Sarvi
    =20
    =20
               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Dave Burke
               Sent: Monday, August 22, 2005 10:45 PM
               To: Shanmugham, Saravanan; Brett Gavagni;=20
     speechsc@ietf.org
               Subject: Re: [Speechsc] MRCPv2 requirements for VoiceXML
               "inputmodes" property
    =20
               Sarvi,
    =20
               The changes below look mostly fine to me with the bug
               noted (point C in
              =20
     http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
               1485.html).
    =20
               One concern I do have with it is that -07 says about the
               speechrecog
               resource:
    =20
               "This resource generates barge-in events for
                  speech.  By analyzing the grammars that are
          activated by the
                  RECOGNIZE method, it also figures out if a barge-in
               should occur for
                  DTMF or not."
    =20
               whereas I thought it ought to say (based on our=20
     discussions):
    =20
                  "This resource generates barge-in events for
                  speech and/or DTMF.  By analyzing the grammar types
               that are activated
                  by the RECOGNIZE method, it determines  if a barge-in
               should occur for
                  speech and/or DTMF"
    =20
               The concern I have with the current wording is=20
     that during
               a call, it is common to go from speech+dtmf reco to
               dtmf-only reco and then back to
               speech+dtmf. With current wording, one would need to
               perform a REINVITE
               to change from a speechrecog to a dtmfrecog. To revert to
               speech+dtmf, another REINVITEs is needed. So 2 REINVITEs
               (=3D6 SIP messages) would be needed to handle the common
               use-case for DTMF-only fallback.
    =20
               Dave
    =20
               ----- Original Message -----
               From: "Shanmugham, Saravanan" <sarvi@cisco.com>
               To: "Brett Gavagni" <gavagni@us.ibm.com>;=20
     <speechsc@ietf.org>
               Sent: Monday, August 22, 2005 6:25 PM
               Subject: RE: [Speechsc] MRCPv2 requirements for VoiceXML
               "inputmodes"
               property
    =20
    =20
               Status in Issue Tracker.
               ---------------------------
                Addressed this by changing START-OF-SPEECH to=20
     START-OF-INPUT.
               Added an Input-Type header to all input processing
          resources, with
               values
               "speech" and "dtmf"
               Add text to explain that what constitutes a=20
     barge-in event
               based on the
               resource
               type and the grammars that are active.
          START-OF-INPUT event is
               generated with
               this barge-in condition is met.
    =20
    =20
               Sarvi
    =20
    =20
    =20
                    -----Original Message-----
                    From: speechsc-bounces@ietf.org
                    [mailto:speechsc-bounces@ietf.org] On Behalf Of
          Brett Gavagni
                    Sent: Monday, August 22, 2005 7:53 AM
                    To: speechsc@ietf.org
                    Subject: [Speechsc] MRCPv2 requirements for VoiceXML
                    "inputmodes" property
    =20
                    Hi,
    =20
                    What's the status of this request that was=20
     made a few
                    months ago, as it doesn't appear to have been
          addressed in
                    the -07 draft?
    =20
                    Here's the thread on the topic:
    =20
          http://www1.ietf.org/mail-archive/web/speechsc/current/msg0
                    1234.html
    =20
                    Thanks,
    =20
                    Brett Gavagni
                    WebSphere Voice Server Development
                   =20
     http://www-306.ibm.com/software/pervasive/voice_server/
                    gavagni@us.ibm.com
    =20
    =20
                    _______________________________________________
                    Speechsc mailing list
                    Speechsc@ietf.org
                    https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
               _______________________________________________
               Speechsc mailing list
               Speechsc@ietf.org
               https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
               _______________________________________________
               Speechsc mailing list
               Speechsc@ietf.org
               https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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From: "Dave Burke" <david.burke@voxpilot.com>
To: "Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>, <sarvi@cisco.com>,
	<dburnett@vocalocity.net>
References: <BBF29C9B95E52E4DB5C29A0ACC94E83B016AA1D3@ac-exch1.eu.scansoft.com>
Subject: Re: [Speechsc] Problem with DTMF type-ahead 
Date: Tue, 23 Aug 2005 20:29:10 +0100
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Problem with DTMF type-aheadNice catch. This is a must fix in my =
opinion.

Here's my thought on a resolution:

By default, the recognizer (dtmfrecog or speechrecog) should buffer =
DTMF. This means that if some DTMF digits are entered outside of the =
Recognizing State and a RECOGNIZE request made later, the buffered =
digits and their timings are applied to that recognition. To clear the =
buffer, a header can be added to RECOGNIZE called Clear-DTMF-Buffer: =
true. Any DTMF digits buffered up to that point are discarded.=20

Can supply the text changes if that's helpful (also note small =
refactoring needed in last sentence of section 9.9 regarding not =
buffering anything before RECOGNIZE).=20

Dave


  ----- Original Message -----=20
  From: Reifenrath, Klaus=20
  To: 'sarvi@cisco.com' ; 'dburnett@vocalocity.net'=20
  Cc: 'speechsc@ietf.org'=20
  Sent: Tuesday, August 23, 2005 4:41 PM
  Subject: [Speechsc] Problem with DTMF type-ahead=20


  The MRCP and MRCPv2 specifications define how DTMF input should be =
handled: DTMF input is collected after the RECOGNIZE request is received =
until a term-char or timeout is reached. Then a result or no-match is =
returned based on any active DTMF grammars. The problem is there can be =
a significant amount of time between when one result is processed by the =
application and the next grammar is activated. During that time any DTMF =
input received will be lost. Such DTMF type-ahead is common for frequent =
users of IVR systems, and buffering that type ahead is required by =
VoiceXML 2.0 (see section 4.1.8). Can this requirement of VoiceXML be =
implemented using MRCP (assuming that the media source CANNOT buffer =
DTMF)? If so, how is it done?=20

  Thanks,
  Klaus=20



-------------------------------------------------------------------------=
-----


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

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Problem with DTMF type-ahead</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Nice catch. This is a must fix in my=20
opinion.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Here's my thought on&nbsp;a=20
resolution:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>By default, the recognizer (dtmfrecog =
or=20
speechrecog) should buffer DTMF. This means that if some DTMF digits are =

entered&nbsp;outside of the Recognizing State and a RECOGNIZE request =
made=20
later, the buffered digits and their timings are applied to that =
recognition. To=20
clear the buffer, a header can be added to RECOGNIZE called =
Clear-DTMF-Buffer:=20
true.&nbsp;Any DTMF digits buffered up to that point are discarded.=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Can supply the text changes if that's =
helpful (also=20
note small refactoring needed in last sentence of section 9.9 regarding =
not=20
buffering anything before RECOGNIZE). </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Dave</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3DKlaus.Reifenrath@Scansoft.com=20
  href=3D"mailto:Klaus.Reifenrath@Scansoft.com">Reifenrath, Klaus</A> =
</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A title=3Dsarvi@cisco.com=20
  href=3D"mailto:'sarvi@cisco.com'">'sarvi@cisco.com'</A> ; <A=20
  title=3Ddburnett@vocalocity.net=20
  =
href=3D"mailto:'dburnett@vocalocity.net'">'dburnett@vocalocity.net'</A> =
</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A =
title=3Dspeechsc@ietf.org=20
  href=3D"mailto:'speechsc@ietf.org'">'speechsc@ietf.org'</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, August 23, 2005 =
4:41=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Speechsc] Problem =
with DTMF=20
  type-ahead </DIV>
  <DIV><BR></DIV>
  <P><FONT face=3DArial size=3D2>The MRCP and MRCPv2 specifications =
define how DTMF=20
  input should be handled: DTMF input is collected after the RECOGNIZE =
request=20
  is received until a term-char or timeout is reached. Then a result or =
no-match=20
  is returned based on any active DTMF grammars. The problem is there =
can be a=20
  significant amount of time between when one result is processed by the =

  application and the next grammar is activated. During that time any =
DTMF input=20
  received will be lost. Such DTMF type-ahead is common for frequent =
users of=20
  IVR systems, and buffering that type ahead is required by VoiceXML 2.0 =
(see=20
  section 4.1.8).</FONT><FONT face=3D"Times New Roman"> Can</FONT><FONT =
face=3DArial=20
  size=3D2> this requirement of VoiceXML be implemented using MRCP =
(assuming that=20
  the media source CANNOT buffer DTMF)? If so, how is it done? =
</FONT></P>
  <P><FONT face=3DArial size=3D2>Thanks,<BR>Klaus</FONT> </P>
  <P>
  <HR>

  <P></P>_______________________________________________<BR>Speechsc =
mailing=20
  =
list<BR>Speechsc@ietf.org<BR>https://www1.ietf.org/mailman/listinfo/speec=
hsc<BR></BLOCKQUOTE></BODY></HTML>

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

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Subject: RE: [Speechsc] Problem with DTMF type-ahead 
Date: Tue, 23 Aug 2005 16:35:56 -0400
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Thread-Topic: [Speechsc] Problem with DTMF type-ahead 
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From: "Pierre Forgues" <forgues@nuance.com>
To: "Dave Burke" <david.burke@voxpilot.com>,
	"Klaus Reifenrath" <Klaus.Reifenrath@Scansoft.com>, <sarvi@cisco.com>, 
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We agree with this position. =20

=20

We have implemented dtmf type ahead in our product and adding a header
to give the application the choice to clear the buffer on the next
RECOGNIZE but by default operating on the dtmf buffer is the right
decision.  /Pierre

=20

________________________________

From: speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] On
Behalf Of Dave Burke
Sent: Tuesday, August 23, 2005 3:29 PM
To: Klaus Reifenrath; sarvi@cisco.com; dburnett@vocalocity.net
Cc: speechsc@ietf.org
Subject: Re: [Speechsc] Problem with DTMF type-ahead=20

=20

Nice catch. This is a must fix in my opinion.

=20

Here's my thought on a resolution:

=20

By default, the recognizer (dtmfrecog or speechrecog) should buffer
DTMF. This means that if some DTMF digits are entered outside of the
Recognizing State and a RECOGNIZE request made later, the buffered
digits and their timings are applied to that recognition. To clear the
buffer, a header can be added to RECOGNIZE called Clear-DTMF-Buffer:
true. Any DTMF digits buffered up to that point are discarded.=20

=20

Can supply the text changes if that's helpful (also note small
refactoring needed in last sentence of section 9.9 regarding not
buffering anything before RECOGNIZE).=20

=20

Dave

=20

=20

	----- Original Message -----=20

	From: Reifenrath, Klaus <mailto:Klaus.Reifenrath@Scansoft.com> =20

	To: 'sarvi@cisco.com' ; 'dburnett@vocalocity.net'=20

	Cc: 'speechsc@ietf.org'=20

	Sent: Tuesday, August 23, 2005 4:41 PM

	Subject: [Speechsc] Problem with DTMF type-ahead=20

	=20

	The MRCP and MRCPv2 specifications define how DTMF input should
be handled: DTMF input is collected after the RECOGNIZE request is
received until a term-char or timeout is reached. Then a result or
no-match is returned based on any active DTMF grammars. The problem is
there can be a significant amount of time between when one result is
processed by the application and the next grammar is activated. During
that time any DTMF input received will be lost. Such DTMF type-ahead is
common for frequent users of IVR systems, and buffering that type ahead
is required by VoiceXML 2.0 (see section 4.1.8). Can this requirement of
VoiceXML be implemented using MRCP (assuming that the media source
CANNOT buffer DTMF)? If so, how is it done?=20

	Thanks,
	Klaus=20

=09
________________________________


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


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>We agree with this position. =
&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>We have implemented dtmf type ahead =
in our
product and adding a header to give the application the choice to clear =
the
buffer on the next RECOGNIZE but by default operating on the dtmf buffer =
is the
right decision. &nbsp;/<st1:City w:st=3D"on"><st1:place =
w:st=3D"on">Pierre</st1:place></st1:City><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Dave Burke<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, August 23, =
2005
3:29 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> <st1:PersonName =
w:st=3D"on">Klaus
 Reifenrath</st1:PersonName>; sarvi@cisco.com; =
dburnett@vocalocity.net<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> speechsc@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Speechsc] =
Problem
with DTMF type-ahead </span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Nice catch. This is a must fix in my =
opinion.</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Here's my thought on&nbsp;a =
resolution:</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>By default, the recognizer (dtmfrecog or speechrecog) =
should
buffer DTMF. This means that if some DTMF digits are =
entered&nbsp;outside of
the <st1:place w:st=3D"on"><st1:PlaceName =
w:st=3D"on">Recognizing</st1:PlaceName> <st1:PlaceType
 w:st=3D"on">State</st1:PlaceType></st1:place> and a RECOGNIZE request =
made
later, the buffered digits and their timings are applied to that =
recognition.
To clear the buffer, a header can be added to RECOGNIZE called
Clear-DTMF-Buffer: true.&nbsp;Any DTMF digits buffered up to that point =
are discarded.
</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Can supply the text changes if that's helpful (also =
note
small refactoring needed in last sentence of section 9.9 regarding not
buffering anything before RECOGNIZE). </span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Dave</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<blockquote style=3D'border:none;border-left:solid black =
1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'=
>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>----- Original Message ----- =
<o:p></o:p></span></font></p>

</div>

<div style=3D'font-color:black'>

<p class=3DMsoNormal style=3D'background:#E4E4E4'><b><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;font-weight:bold'>From:</span=
></font></b><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> <a
href=3D"mailto:Klaus.Reifenrath@Scansoft.com"
title=3D"Klaus.Reifenrath@Scansoft.com">Reifenrath, Klaus</a> =
<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;font-weight:bold'>To:</span></font></b><font size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> <a
href=3D"mailto:'sarvi@cisco.com'" =
title=3D"sarvi@cisco.com">'sarvi@cisco.com'</a> ;
<a href=3D"mailto:'dburnett@vocalocity.net'" =
title=3D"dburnett@vocalocity.net">'dburnett@vocalocity.net'</a>
<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;font-weight:bold'>Cc:</span></font></b><font size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> <a
href=3D"mailto:'speechsc@ietf.org'" =
title=3D"speechsc@ietf.org">'speechsc@ietf.org'</a>
<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;font-weight:bold'>Sent:</span></font></b><font =
size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> =
Tuesday, August
23, 2005 4:41 PM<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;font-weight:bold'>Subject:</span></font></b><font =
size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> =
[Speechsc] Problem
with DTMF type-ahead <o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<p><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>The
MRCP and MRCPv2 specifications define how DTMF input should be handled: =
DTMF
input is collected after the RECOGNIZE request is received until a =
term-char or
timeout is reached. Then a result or no-match is returned based on any =
active
DTMF grammars. The problem is there can be a significant amount of time =
between
when one result is processed by the application and the next grammar is
activated. During that time any DTMF input received will be lost. Such =
DTMF
type-ahead is common for frequent users of IVR systems, and buffering =
that type
ahead is required by VoiceXML 2.0 (see section 4.1.8).</span></font> =
Can<font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> this
requirement of VoiceXML be implemented using MRCP (assuming that the =
media
source CANNOT buffer DTMF)? If so, how is it done? =
</span></font><o:p></o:p></p>

<p><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Thanks,<br>
Klaus</span></font> <o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>_______________________________________________<br>
Speechsc mailing list<br>
Speechsc@ietf.org<br>
https://www1.ietf.org/mailman/listinfo/speechsc<o:p></o:p></span></font><=
/p>

</blockquote>

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Thread-Topic: Problem with DTMF type-ahead 
Thread-Index: AcWn+6tjuWWuWeMTTXKPwEhMTCB1xgABfb3w
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>,
	<dburnett@vocalocity.net>
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Subject: [Speechsc] RE: Problem with DTMF type-ahead 
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=20
I had typed this response out earlier but wanted to see if there was any
support for this capability. So sending it out now.
=20
The solution I would suggest is a new header=20
           Type-Ahead: <time-in-milli-seconds>=20
header that can be sent on the SET-PARAMS, GET-PARAMS and RECOGNIZE
methods.
=20
default is 0 which is todays no buffering case. If set to some value,
the resource is expected to buffer upto <time-in-milliseconds> of audio
and use it during the recognize as a type-ahead buffer. I can see this
being a boolean field as well, but the <time-in-milliseconds> approach
is little more flexible.
=20
Sarvi


________________________________

	From: Reifenrath, Klaus [mailto:Klaus.Reifenrath@Scansoft.com]=20
	Sent: Tuesday, August 23, 2005 8:41 AM
	To: Shanmugham, Saravanan; 'dburnett@vocalocity.net'
	Cc: 'speechsc@ietf.org'
	Subject: Problem with DTMF type-ahead=20
=09
=09

	The MRCP and MRCPv2 specifications define how DTMF input should
be handled: DTMF input is collected after the RECOGNIZE request is
received until a term-char or timeout is reached. Then a result or
no-match is returned based on any active DTMF grammars. The problem is
there can be a significant amount of time between when one result is
processed by the application and the next grammar is activated. During
that time any DTMF input received will be lost. Such DTMF type-ahead is
common for frequent users of IVR systems, and buffering that type ahead
is required by VoiceXML 2.0 (see section 4.1.8). Can this requirement of
VoiceXML be implemented using MRCP (assuming that the media source
CANNOT buffer DTMF)? If so, how is it done?=20

	Thanks,
	Klaus=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Problem with DTMF type-ahead</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1515" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>I had typed this response out earlier but =
wanted to see=20
if there was any support for this capability. So sending it out=20
now.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>The solution I would suggest is a new header=20
</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
Type-Ahead: &lt;time-in-milli-seconds&gt; </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>header that can be sent on the SET-PARAMS, =
GET-PARAMS=20
and RECOGNIZE methods.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>default is 0 which is todays no buffering =
case. If set=20
to some value, the resource is expected to buffer upto=20
&lt;time-in-milliseconds&gt; of audio and use it during the recognize as =
a=20
type-ahead buffer. I can see this being a boolean field as well, but the =

&lt;time-in-milliseconds&gt; approach is little more=20
flexible.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>Sarvi</SPAN></FONT></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Reifenrath, Klaus=20
  [mailto:Klaus.Reifenrath@Scansoft.com] <BR><B>Sent:</B> Tuesday, =
August 23,=20
  2005 8:41 AM<BR><B>To:</B> Shanmugham, Saravanan;=20
  'dburnett@vocalocity.net'<BR><B>Cc:</B> =
'speechsc@ietf.org'<BR><B>Subject:</B>=20
  Problem with DTMF type-ahead <BR></FONT><BR></DIV>
  <DIV></DIV>
  <P><FONT face=3DArial size=3D2>The MRCP and MRCPv2 specifications =
define how DTMF=20
  input should be handled: DTMF input is collected after the RECOGNIZE =
request=20
  is received until a term-char or timeout is reached. Then a result or =
no-match=20
  is returned based on any active DTMF grammars. The problem is there =
can be a=20
  significant amount of time between when one result is processed by the =

  application and the next grammar is activated. During that time any =
DTMF input=20
  received will be lost. Such DTMF type-ahead is common for frequent =
users of=20
  IVR systems, and buffering that type ahead is required by VoiceXML 2.0 =
(see=20
  section 4.1.8).</FONT><FONT face=3D"Times New Roman"> Can</FONT><FONT =
face=3DArial=20
  size=3D2> this requirement of VoiceXML be implemented using MRCP =
(assuming that=20
  the media source CANNOT buffer DTMF)? If so, how is it done? =
</FONT></P>
  <P><FONT face=3DArial size=3D2>Thanks,<BR>Klaus</FONT>=20
</P></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
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From speechsc-bounces@ietf.org Tue Aug 23 17:24:21 2005
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Subject: RE: [Speechsc] Recognition Completion-Cause Codes 003 and 008
	revisited
Date: Tue, 23 Aug 2005 14:21:38 -0700
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Thread-Topic: [Speechsc] Recognition Completion-Cause Codes 003 and 008
	revisited
Thread-Index: AcWn8DBSk53rlDwHSgW+Bg7zWDr3zQAN+GFw
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Andrew Wahbe" <awahbe@voicegenie.com>, <speechsc@ietf.org>
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I was going to remove Recognition-Timeout based on the thread on the
confusion between this cause and max-speech-timeout etc.

But it was agreed we need a timeout capability for HotWord Recognition
based ona separate thread.

In the hotword case, none of the timeout desciptions match the way
timeout happens in hotword recognition. Since there, even if the words
spoken until that point did not match any of the hotwords, the
recognition would still continue. Hence a separate completion code for
hotword recognition timeout was needed. And I have used 03 for that use
case.

Sarvi

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Andrew Wahbe
     Sent: Tuesday, August 23, 2005 7:34 AM
     To: speechsc@ietf.org
     Subject: [Speechsc] Recognition Completion-Cause Codes 003=20
     and 008 revisited
    =20
     A few weeks ago, there was an extensive discussion about=20
     the recognition completion-cause codes for cases where=20
     recognition-timers fire. There seemed to be two codes, 003=20
     and 008, that meant the same thing -- the recognition was=20
     aborted due to the recognition-timer firing. This case=20
     maps to the maxspeechtimeout event in VoiceXML.
    =20
     A lot of hypothesizing was done to try and figure out what=20
     the difference between the codes was. After some=20
     discussion it seems we ended up with no codes that map to=20
     maxspeechtimeout in VoiceXML (i.e.=20
     the recognition was aborted because of recognition-timeout=20
     and there is no match).
    =20
     I think things would be set right if 003 was changed from:
    =20
        | 003    | recognition-timeout   | RECOGNIZE in hotword=20
     mode        |
        |        |                       | completed without a=20
     match due to |
        |        |                       | a=20
     recognition-timeout            |
    =20
     to:
    =20
        | 003    | recognition-timeout   | RECOGNIZE completed=20
     without a    |
        |        |                       | match due to a      =20
                 |
        |        |                       | recognition-timeout =20
                 |
    =20
    =20
     I am not sure why a special code is needed for this timer=20
     firing in hotword mode. Am I missing something?
    =20
     Andrew Wahbe
     VoiceGenie Technologies
    =20

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



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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead 
Date: Tue, 23 Aug 2005 18:19:44 -0400
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Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead 
Thread-Index: AcWn+6tjuWWuWeMTTXKPwEhMTCB1xgABfb3wAAu+kHA=
From: "Jeff Haynie" <jhaynie@vocalocity.net>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>,
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It would seem like we're not trying to buffer audio (generally) but only
the DIGIT values themselves. =20

It would seem better to just be a header such as:
=20
DTMF-Buffer: <true|false>
=20
instead.
=20
Jeff

________________________________

From: speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] On
Behalf Of Shanmugham, Saravanan
Sent: Tuesday, August 23, 2005 4:59 PM
To: Reifenrath, Klaus; dburnett@vocalocity.net
Cc: speechsc@ietf.org
Subject: [Speechsc] RE: Problem with DTMF type-ahead=20


=20
I had typed this response out earlier but wanted to see if there was any
support for this capability. So sending it out now.
=20
The solution I would suggest is a new header=20
           Type-Ahead: <time-in-milli-seconds>=20
header that can be sent on the SET-PARAMS, GET-PARAMS and RECOGNIZE
methods.
=20
default is 0 which is todays no buffering case. If set to some value,
the resource is expected to buffer upto <time-in-milliseconds> of audio
and use it during the recognize as a type-ahead buffer. I can see this
being a boolean field as well, but the <time-in-milliseconds> approach
is little more flexible.
=20
Sarvi


________________________________

	From: Reifenrath, Klaus [mailto:Klaus.Reifenrath@Scansoft.com]=20
	Sent: Tuesday, August 23, 2005 8:41 AM
	To: Shanmugham, Saravanan; 'dburnett@vocalocity.net'
	Cc: 'speechsc@ietf.org'
	Subject: Problem with DTMF type-ahead=20
=09
=09

	The MRCP and MRCPv2 specifications define how DTMF input should
be handled: DTMF input is collected after the RECOGNIZE request is
received until a term-char or timeout is reached. Then a result or
no-match is returned based on any active DTMF grammars. The problem is
there can be a significant amount of time between when one result is
processed by the application and the next grammar is activated. During
that time any DTMF input received will be lost. Such DTMF type-ahead is
common for frequent users of IVR systems, and buffering that type ahead
is required by VoiceXML 2.0 (see section 4.1.8). Can this requirement of
VoiceXML be implemented using MRCP (assuming that the media source
CANNOT buffer DTMF)? If so, how is it done?=20

	Thanks,
	Klaus=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D929381822-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>It would seem like we're not trying to buffer =
audio=20
(generally) but only the DIGIT values themselves.&nbsp;=20
</FONT></SPAN></DIV><SPAN class=3D929381822-23082005>
<DIV dir=3Dltr align=3Dleft><BR><FONT face=3DArial color=3D#0000ff =
size=3D2>It would seem=20
better to just be a header such as:</FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft></SPAN><SPAN =
class=3D929381822-23082005><FONT face=3DArial=20
color=3D#0000ff size=3D2>DTMF-Buffer: =
&lt;true|false&gt;</FONT></SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D929381822-23082005></SPAN><FONT face=3DArial><FONT=20
color=3D#0000ff><FONT size=3D2>i<SPAN=20
class=3D929381822-23082005>nstead.</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D929381822-23082005></SPAN></FONT></FONT></FONT><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D929381822-23082005><FONT face=3DArial color=3D#0000ff =

size=3D2>Jeff</FONT></SPAN></DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> speechsc-bounces@ietf.org=20
[mailto:speechsc-bounces@ietf.org] <B>On Behalf Of </B>Shanmugham,=20
Saravanan<BR><B>Sent:</B> Tuesday, August 23, 2005 4:59 PM<BR><B>To:</B> =

Reifenrath, Klaus; dburnett@vocalocity.net<BR><B>Cc:</B>=20
speechsc@ietf.org<BR><B>Subject:</B> [Speechsc] RE: Problem with DTMF =
type-ahead=20
<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>I had typed this response out earlier but =
wanted to see=20
if there was any support for this capability. So sending it out=20
now.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>The solution I would suggest is a new header=20
</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
Type-Ahead: &lt;time-in-milli-seconds&gt; </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>header that can be sent on the SET-PARAMS, =
GET-PARAMS=20
and RECOGNIZE methods.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>default is 0 which is todays no buffering =
case. If set=20
to some value, the resource is expected to buffer upto=20
&lt;time-in-milliseconds&gt; of audio and use it during the recognize as =
a=20
type-ahead buffer. I can see this being a boolean field as well, but the =

&lt;time-in-milliseconds&gt; approach is little more=20
flexible.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D256224216-23082005>Sarvi</SPAN></FONT></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Reifenrath, Klaus=20
  [mailto:Klaus.Reifenrath@Scansoft.com] <BR><B>Sent:</B> Tuesday, =
August 23,=20
  2005 8:41 AM<BR><B>To:</B> Shanmugham, Saravanan;=20
  'dburnett@vocalocity.net'<BR><B>Cc:</B> =
'speechsc@ietf.org'<BR><B>Subject:</B>=20
  Problem with DTMF type-ahead <BR></FONT><BR></DIV>
  <DIV></DIV>
  <P><FONT face=3DArial size=3D2>The MRCP and MRCPv2 specifications =
define how DTMF=20
  input should be handled: DTMF input is collected after the RECOGNIZE =
request=20
  is received until a term-char or timeout is reached. Then a result or =
no-match=20
  is returned based on any active DTMF grammars. The problem is there =
can be a=20
  significant amount of time between when one result is processed by the =

  application and the next grammar is activated. During that time any =
DTMF input=20
  received will be lost. Such DTMF type-ahead is common for frequent =
users of=20
  IVR systems, and buffering that type ahead is required by VoiceXML 2.0 =
(see=20
  section 4.1.8).</FONT><FONT face=3D"Times New Roman"> Can</FONT><FONT =
face=3DArial=20
  size=3D2> this requirement of VoiceXML be implemented using MRCP =
(assuming that=20
  the media source CANNOT buffer DTMF)? If so, how is it done? =
</FONT></P>
  <P><FONT face=3DArial size=3D2>Thanks,<BR>Klaus</FONT>=20
</P></BLOCKQUOTE></BODY></HTML>

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From speechsc-bounces@ietf.org Tue Aug 23 18:31:01 2005
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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead 
Date: Tue, 23 Aug 2005 15:30:39 -0700
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Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead 
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From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Jeff Haynie" <jhaynie@vocalocity.net>,
	"Reifenrath, Klaus" <Klaus.Reifenrath@Scansoft.com>,
	<dburnett@vocalocity.net>
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Are we assuming then that this is only limited to DTMF type-ahead and is
not speak ahead as well?
=20
Sarvi


________________________________

	From: Jeff Haynie [mailto:jhaynie@vocalocity.net]=20
	Sent: Tuesday, August 23, 2005 3:20 PM
	To: Shanmugham, Saravanan; Reifenrath, Klaus;
dburnett@vocalocity.net
	Cc: speechsc@ietf.org
	Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead=20
=09
=09
	It would seem like we're not trying to buffer audio (generally)
but only the DIGIT values themselves. =20
=09

	It would seem better to just be a header such as:
	=20
	DTMF-Buffer: <true|false>
	=20
	instead.
	=20
	Jeff

________________________________

	From: speechsc-bounces@ietf.org
[mailto:speechsc-bounces@ietf.org] On Behalf Of Shanmugham, Saravanan
	Sent: Tuesday, August 23, 2005 4:59 PM
	To: Reifenrath, Klaus; dburnett@vocalocity.net
	Cc: speechsc@ietf.org
	Subject: [Speechsc] RE: Problem with DTMF type-ahead=20
=09
=09
	=20
	I had typed this response out earlier but wanted to see if there
was any support for this capability. So sending it out now.
	=20
	The solution I would suggest is a new header=20
	           Type-Ahead: <time-in-milli-seconds>=20
	header that can be sent on the SET-PARAMS, GET-PARAMS and
RECOGNIZE methods.
	=20
	default is 0 which is todays no buffering case. If set to some
value, the resource is expected to buffer upto <time-in-milliseconds> of
audio and use it during the recognize as a type-ahead buffer. I can see
this being a boolean field as well, but the <time-in-milliseconds>
approach is little more flexible.
	=20
	Sarvi


________________________________

		From: Reifenrath, Klaus
[mailto:Klaus.Reifenrath@Scansoft.com]=20
		Sent: Tuesday, August 23, 2005 8:41 AM
		To: Shanmugham, Saravanan; 'dburnett@vocalocity.net'
		Cc: 'speechsc@ietf.org'
		Subject: Problem with DTMF type-ahead=20
	=09
	=09

		The MRCP and MRCPv2 specifications define how DTMF input
should be handled: DTMF input is collected after the RECOGNIZE request
is received until a term-char or timeout is reached. Then a result or
no-match is returned based on any active DTMF grammars. The problem is
there can be a significant amount of time between when one result is
processed by the application and the next grammar is activated. During
that time any DTMF input received will be lost. Such DTMF type-ahead is
common for frequent users of IVR systems, and buffering that type ahead
is required by VoiceXML 2.0 (see section 4.1.8). Can this requirement of
VoiceXML be implemented using MRCP (assuming that the media source
CANNOT buffer DTMF)? If so, how is it done?=20

		Thanks,
		Klaus=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Problem with DTMF type-ahead</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1515" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D531192922-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Are we assuming then that this is =
only&nbsp;limited=20
to&nbsp;DTMF type-ahead and is not speak ahead as =
well?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D531192922-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D531192922-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Sarvi</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Jeff Haynie=20
  [mailto:jhaynie@vocalocity.net] <BR><B>Sent:</B> Tuesday, August 23, =
2005 3:20=20
  PM<BR><B>To:</B> Shanmugham, Saravanan; Reifenrath, Klaus;=20
  dburnett@vocalocity.net<BR><B>Cc:</B> =
speechsc@ietf.org<BR><B>Subject:</B> RE:=20
  [Speechsc] RE: Problem with DTMF type-ahead <BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D929381822-23082005><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>It would seem like we're not trying to buffer =
audio=20
  (generally) but only the DIGIT values themselves.&nbsp;=20
  </FONT></SPAN></DIV><SPAN class=3D929381822-23082005>
  <DIV dir=3Dltr align=3Dleft><BR><FONT face=3DArial color=3D#0000ff =
size=3D2>It would=20
  seem better to just be a header such as:</FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft></SPAN><SPAN =
class=3D929381822-23082005><FONT face=3DArial=20
  color=3D#0000ff size=3D2>DTMF-Buffer: =
&lt;true|false&gt;</FONT></SPAN></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D929381822-23082005></SPAN><FONT face=3DArial><FONT=20
  color=3D#0000ff><FONT size=3D2>i<SPAN=20
  class=3D929381822-23082005>nstead.</SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
  class=3D929381822-23082005></SPAN></FONT></FONT></FONT><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D929381822-23082005><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Jeff</FONT></SPAN></DIV>
  <DIV><BR></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> speechsc-bounces@ietf.org=20
  [mailto:speechsc-bounces@ietf.org] <B>On Behalf Of </B>Shanmugham,=20
  Saravanan<BR><B>Sent:</B> Tuesday, August 23, 2005 4:59 =
PM<BR><B>To:</B>=20
  Reifenrath, Klaus; dburnett@vocalocity.net<BR><B>Cc:</B>=20
  speechsc@ietf.org<BR><B>Subject:</B> [Speechsc] RE: Problem with DTMF=20
  type-ahead <BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>I had typed this response out earlier but =
wanted to=20
  see if there was any support for this capability. So sending it out=20
  now.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>The solution I would suggest is a new =
header=20
  </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  =
class=3D256224216-23082005>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
  Type-Ahead: &lt;time-in-milli-seconds&gt; </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>header that can be sent on the SET-PARAMS, =
GET-PARAMS=20
  and RECOGNIZE methods.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>default is 0 which is todays no buffering =
case. If=20
  set to some value, the resource is expected to buffer upto=20
  &lt;time-in-milliseconds&gt; of audio and use it during the recognize =
as a=20
  type-ahead buffer. I can see this being a boolean field as well, but =
the=20
  &lt;time-in-milliseconds&gt; approach is little more=20
  flexible.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>Sarvi</SPAN></FONT></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> Reifenrath, Klaus=20
    [mailto:Klaus.Reifenrath@Scansoft.com] <BR><B>Sent:</B> Tuesday, =
August 23,=20
    2005 8:41 AM<BR><B>To:</B> Shanmugham, Saravanan;=20
    'dburnett@vocalocity.net'<BR><B>Cc:</B>=20
    'speechsc@ietf.org'<BR><B>Subject:</B> Problem with DTMF type-ahead=20
    <BR></FONT><BR></DIV>
    <DIV></DIV>
    <P><FONT face=3DArial size=3D2>The MRCP and MRCPv2 specifications =
define how=20
    DTMF input should be handled: DTMF input is collected after the =
RECOGNIZE=20
    request is received until a term-char or timeout is reached. Then a =
result=20
    or no-match is returned based on any active DTMF grammars. The =
problem is=20
    there can be a significant amount of time between when one result is =

    processed by the application and the next grammar is activated. =
During that=20
    time any DTMF input received will be lost. Such DTMF type-ahead is =
common=20
    for frequent users of IVR systems, and buffering that type ahead is =
required=20
    by VoiceXML 2.0 (see section 4.1.8).</FONT><FONT face=3D"Times New =
Roman">=20
    Can</FONT><FONT face=3DArial size=3D2> this requirement of VoiceXML =
be=20
    implemented using MRCP (assuming that the media source CANNOT buffer =
DTMF)?=20
    If so, how is it done? </FONT></P>
    <P><FONT face=3DArial size=3D2>Thanks,<BR>Klaus</FONT>=20
</P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From: Corby Anderson <corby@tellme.com>
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To: speechsc@ietf.org
Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
References: <03772D1EC8DE624A863058C75874A75C2C41E6@vtg-um-e2k6.sj21ad.cisco.com>
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I prefer the flexible "Type-Ahead: <time-in-milliseconds>" approach.  
This would be useful for implementing vxml listen states that happen 
after data tags.  If a data fetch takes longer than several seconds, you 
might not want to play dtmf that is older than some limit.

Corby Anderson
Tellme Networks, Inc.

Shanmugham, Saravanan wrote:

> Are we assuming then that this is only limited to DTMF type-ahead and 
> is not speak ahead as well?
>  
> Sarvi
>
>     ------------------------------------------------------------------------
>     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
>     *Sent:* Tuesday, August 23, 2005 3:20 PM
>     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
>     dburnett@vocalocity.net
>     *Cc:* speechsc@ietf.org
>     *Subject:* RE: [Speechsc] RE: Problem with DTMF type-ahead
>
>     It would seem like we're not trying to buffer audio (generally)
>     but only the DIGIT values themselves. 
>
>     It would seem better to just be a header such as:
>      
>     DTMF-Buffer: <true|false>
>      
>     instead.
>      
>     Jeff
>
>     ------------------------------------------------------------------------
>     *From:* speechsc-bounces@ietf.org
>     [mailto:speechsc-bounces@ietf.org] *On Behalf Of *Shanmugham,
>     Saravanan
>     *Sent:* Tuesday, August 23, 2005 4:59 PM
>     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
>     *Cc:* speechsc@ietf.org
>     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
>
>      
>     I had typed this response out earlier but wanted to see if there
>     was any support for this capability. So sending it out now.
>      
>     The solution I would suggest is a new header
>                Type-Ahead: <time-in-milli-seconds>
>     header that can be sent on the SET-PARAMS, GET-PARAMS and
>     RECOGNIZE methods.
>      
>     default is 0 which is todays no buffering case. If set to some
>     value, the resource is expected to buffer upto
>     <time-in-milliseconds> of audio and use it during the recognize as
>     a type-ahead buffer. I can see this being a boolean field as well,
>     but the <time-in-milliseconds> approach is little more flexible.
>      
>     Sarvi
>
>         ------------------------------------------------------------------------
>         *From:* Reifenrath, Klaus [mailto:Klaus.Reifenrath@Scansoft.com]
>         *Sent:* Tuesday, August 23, 2005 8:41 AM
>         *To:* Shanmugham, Saravanan; 'dburnett@vocalocity.net'
>         *Cc:* 'speechsc@ietf.org'
>         *Subject:* Problem with DTMF type-ahead
>
>         The MRCP and MRCPv2 specifications define how DTMF input
>         should be handled: DTMF input is collected after the RECOGNIZE
>         request is received until a term-char or timeout is reached.
>         Then a result or no-match is returned based on any active DTMF
>         grammars. The problem is there can be a significant amount of
>         time between when one result is processed by the application
>         and the next grammar is activated. During that time any DTMF
>         input received will be lost. Such DTMF type-ahead is common
>         for frequent users of IVR systems, and buffering that type
>         ahead is required by VoiceXML 2.0 (see section 4.1.8). Can
>         this requirement of VoiceXML be implemented using MRCP
>         (assuming that the media source CANNOT buffer DTMF)? If so,
>         how is it done?
>
>         Thanks,
>         Klaus
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Speechsc mailing list
>Speechsc@ietf.org
>https://www1.ietf.org/mailman/listinfo/speechsc
>  
>

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



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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead 
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From: "Jeff Haynie" <jhaynie@vocalocity.net>
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Sarvi, et al:
=20
Is there a use case for speak ahead?  I can't honestly think of one ...
maybe Dan or someone else knows of one they could speak too.  It also
seems like that would introduce a lot of other complexity with partial
packets, etc. that you isn't the case with DTMF.
=20
Jeff

________________________________

From: Shanmugham, Saravanan [mailto:sarvi@cisco.com]=20
Sent: Tuesday, August 23, 2005 6:31 PM
To: Jeff Haynie; Reifenrath, Klaus; dburnett@vocalocity.net
Cc: speechsc@ietf.org
Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead=20


Are we assuming then that this is only limited to DTMF type-ahead and is
not speak ahead as well?
=20
Sarvi


________________________________

	From: Jeff Haynie [mailto:jhaynie@vocalocity.net]=20
	Sent: Tuesday, August 23, 2005 3:20 PM
	To: Shanmugham, Saravanan; Reifenrath, Klaus;
dburnett@vocalocity.net
	Cc: speechsc@ietf.org
	Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead=20
=09
=09
	It would seem like we're not trying to buffer audio (generally)
but only the DIGIT values themselves. =20
=09

	It would seem better to just be a header such as:
	=20
	DTMF-Buffer: <true|false>
	=20
	instead.
	=20
	Jeff

________________________________

	From: speechsc-bounces@ietf.org
[mailto:speechsc-bounces@ietf.org] On Behalf Of Shanmugham, Saravanan
	Sent: Tuesday, August 23, 2005 4:59 PM
	To: Reifenrath, Klaus; dburnett@vocalocity.net
	Cc: speechsc@ietf.org
	Subject: [Speechsc] RE: Problem with DTMF type-ahead=20
=09
=09
	=20
	I had typed this response out earlier but wanted to see if there
was any support for this capability. So sending it out now.
	=20
	The solution I would suggest is a new header=20
	           Type-Ahead: <time-in-milli-seconds>=20
	header that can be sent on the SET-PARAMS, GET-PARAMS and
RECOGNIZE methods.
	=20
	default is 0 which is todays no buffering case. If set to some
value, the resource is expected to buffer upto <time-in-milliseconds> of
audio and use it during the recognize as a type-ahead buffer. I can see
this being a boolean field as well, but the <time-in-milliseconds>
approach is little more flexible.
	=20
	Sarvi


________________________________

		From: Reifenrath, Klaus
[mailto:Klaus.Reifenrath@Scansoft.com]=20
		Sent: Tuesday, August 23, 2005 8:41 AM
		To: Shanmugham, Saravanan; 'dburnett@vocalocity.net'
		Cc: 'speechsc@ietf.org'
		Subject: Problem with DTMF type-ahead=20
	=09
	=09

		The MRCP and MRCPv2 specifications define how DTMF input
should be handled: DTMF input is collected after the RECOGNIZE request
is received until a term-char or timeout is reached. Then a result or
no-match is returned based on any active DTMF grammars. The problem is
there can be a significant amount of time between when one result is
processed by the application and the next grammar is activated. During
that time any DTMF input received will be lost. Such DTMF type-ahead is
common for frequent users of IVR systems, and buffering that type ahead
is required by VoiceXML 2.0 (see section 4.1.8). Can this requirement of
VoiceXML be implemented using MRCP (assuming that the media source
CANNOT buffer DTMF)? If so, how is it done?=20

		Thanks,
		Klaus=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Problem with DTMF type-ahead</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2722" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D213160300-24082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Sarvi, et al:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D213160300-24082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D213160300-24082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Is there a use case for speak ahead?&nbsp; I =
can't honestly=20
think of one ... maybe Dan or someone else&nbsp;knows of one they could =
speak=20
too.&nbsp; It also seems like that would introduce a lot of other =
complexity=20
with partial packets, etc. that you isn't the case with=20
DTMF.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D213160300-24082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D213160300-24082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Jeff</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Shanmugham, Saravanan=20
[mailto:sarvi@cisco.com] <BR><B>Sent:</B> Tuesday, August 23, 2005 6:31=20
PM<BR><B>To:</B> Jeff Haynie; Reifenrath, Klaus;=20
dburnett@vocalocity.net<BR><B>Cc:</B> =
speechsc@ietf.org<BR><B>Subject:</B> RE:=20
[Speechsc] RE: Problem with DTMF type-ahead <BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D531192922-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Are we assuming then that this is =
only&nbsp;limited=20
to&nbsp;DTMF type-ahead and is not speak ahead as =
well?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D531192922-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D531192922-23082005><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Sarvi</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Jeff Haynie=20
  [mailto:jhaynie@vocalocity.net] <BR><B>Sent:</B> Tuesday, August 23, =
2005 3:20=20
  PM<BR><B>To:</B> Shanmugham, Saravanan; Reifenrath, Klaus;=20
  dburnett@vocalocity.net<BR><B>Cc:</B> =
speechsc@ietf.org<BR><B>Subject:</B> RE:=20
  [Speechsc] RE: Problem with DTMF type-ahead <BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D929381822-23082005><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>It would seem like we're not trying to buffer =
audio=20
  (generally) but only the DIGIT values themselves.&nbsp;=20
  </FONT></SPAN></DIV><SPAN class=3D929381822-23082005>
  <DIV dir=3Dltr align=3Dleft><BR><FONT face=3DArial color=3D#0000ff =
size=3D2>It would=20
  seem better to just be a header such as:</FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft></SPAN><SPAN =
class=3D929381822-23082005><FONT face=3DArial=20
  color=3D#0000ff size=3D2>DTMF-Buffer: =
&lt;true|false&gt;</FONT></SPAN></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D929381822-23082005></SPAN><FONT face=3DArial><FONT=20
  color=3D#0000ff><FONT size=3D2>i<SPAN=20
  class=3D929381822-23082005>nstead.</SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
  class=3D929381822-23082005></SPAN></FONT></FONT></FONT><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D929381822-23082005><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Jeff</FONT></SPAN></DIV>
  <DIV><BR></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> speechsc-bounces@ietf.org=20
  [mailto:speechsc-bounces@ietf.org] <B>On Behalf Of </B>Shanmugham,=20
  Saravanan<BR><B>Sent:</B> Tuesday, August 23, 2005 4:59 =
PM<BR><B>To:</B>=20
  Reifenrath, Klaus; dburnett@vocalocity.net<BR><B>Cc:</B>=20
  speechsc@ietf.org<BR><B>Subject:</B> [Speechsc] RE: Problem with DTMF=20
  type-ahead <BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>I had typed this response out earlier but =
wanted to=20
  see if there was any support for this capability. So sending it out=20
  now.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>The solution I would suggest is a new =
header=20
  </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  =
class=3D256224216-23082005>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
  Type-Ahead: &lt;time-in-milli-seconds&gt; </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>header that can be sent on the SET-PARAMS, =
GET-PARAMS=20
  and RECOGNIZE methods.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>default is 0 which is todays no buffering =
case. If=20
  set to some value, the resource is expected to buffer upto=20
  &lt;time-in-milliseconds&gt; of audio and use it during the recognize =
as a=20
  type-ahead buffer. I can see this being a boolean field as well, but =
the=20
  &lt;time-in-milliseconds&gt; approach is little more=20
  flexible.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D256224216-23082005>Sarvi</SPAN></FONT></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> Reifenrath, Klaus=20
    [mailto:Klaus.Reifenrath@Scansoft.com] <BR><B>Sent:</B> Tuesday, =
August 23,=20
    2005 8:41 AM<BR><B>To:</B> Shanmugham, Saravanan;=20
    'dburnett@vocalocity.net'<BR><B>Cc:</B>=20
    'speechsc@ietf.org'<BR><B>Subject:</B> Problem with DTMF type-ahead=20
    <BR></FONT><BR></DIV>
    <DIV></DIV>
    <P><FONT face=3DArial size=3D2>The MRCP and MRCPv2 specifications =
define how=20
    DTMF input should be handled: DTMF input is collected after the =
RECOGNIZE=20
    request is received until a term-char or timeout is reached. Then a =
result=20
    or no-match is returned based on any active DTMF grammars. The =
problem is=20
    there can be a significant amount of time between when one result is =

    processed by the application and the next grammar is activated. =
During that=20
    time any DTMF input received will be lost. Such DTMF type-ahead is =
common=20
    for frequent users of IVR systems, and buffering that type ahead is =
required=20
    by VoiceXML 2.0 (see section 4.1.8).</FONT><FONT face=3D"Times New =
Roman">=20
    Can</FONT><FONT face=3DArial size=3D2> this requirement of VoiceXML =
be=20
    implemented using MRCP (assuming that the media source CANNOT buffer =
DTMF)?=20
    If so, how is it done? </FONT></P>
    <P><FONT face=3DArial size=3D2>Thanks,<BR>Klaus</FONT>=20
</P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From speechsc-bounces@ietf.org Wed Aug 24 02:15:27 2005
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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
Date: Wed, 24 Aug 2005 02:14:19 -0400
Message-ID: <330A23D8336C0346B5C1A5BB19666647E06607@ATLANTIS.Brooktrout.com>
Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead
Thread-Index: AcWoO0FVkK5nyeW4Sn2qclBDDIBrlwANxhgg
From: "Eric Burger" <eburger@brooktrout.com>
To: <speechsc@ietf.org>
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[not as chair]

I can see something like "Type-Ahead" for SET/GET-PARAMS, which sets a
global digit buffer.  I would suggest a more descriptive name, like
DTMF-Buffer-Time.

That said, I don't see it working for the RECOGNIZE method.  The speech
resource needs to know *before* the RECOGNIZE request that it needs to
buffer DTMF.  If it comes in with the RECOGNIZE request, it will be too
late.

For example:

SET-PARAMS
DTMF-Buffer-Time: 10000
...
<sets global buffer to 10s; platform keeps 10 seconds of DTMF buffered>


RECOGNIZE
DTMF-Buffer-Time: 20000
...
<sets buffer to 20s.  Oops: platform already dropped first 10 seconds
because it was only buffering 10 seconds>



The sensible per-RECOGNIZE attribute is to flush the buffer.  While one
might be tempted to say "DTMF-Buffer-Time: 0" means flush the buffer, I
am sure someone will try to use it with a non-zero value.  I would offer
we use a different attribute to indicate flushing the buffer before
recognition, like "Flush-DTMF: true".

-----Original Message-----
From: speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] On
Behalf Of Corby Anderson
Sent: Tuesday, August 23, 2005 7:34 PM
To: speechsc@ietf.org
Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead

I prefer the flexible "Type-Ahead: <time-in-milliseconds>" approach. =20
This would be useful for implementing vxml listen states that happen=20
after data tags.  If a data fetch takes longer than several seconds, you

might not want to play dtmf that is older than some limit.

Corby Anderson
Tellme Networks, Inc.

Shanmugham, Saravanan wrote:

> Are we assuming then that this is only limited to DTMF type-ahead and=20
> is not speak ahead as well?
> =20
> Sarvi
>
>
------------------------------------------------------------------------
>     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
>     *Sent:* Tuesday, August 23, 2005 3:20 PM
>     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
>     dburnett@vocalocity.net
>     *Cc:* speechsc@ietf.org
>     *Subject:* RE: [Speechsc] RE: Problem with DTMF type-ahead
>
>     It would seem like we're not trying to buffer audio (generally)
>     but only the DIGIT values themselves.=20
>
>     It would seem better to just be a header such as:
>     =20
>     DTMF-Buffer: <true|false>
>     =20
>     instead.
>     =20
>     Jeff
>
>
------------------------------------------------------------------------
>     *From:* speechsc-bounces@ietf.org
>     [mailto:speechsc-bounces@ietf.org] *On Behalf Of *Shanmugham,
>     Saravanan
>     *Sent:* Tuesday, August 23, 2005 4:59 PM
>     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
>     *Cc:* speechsc@ietf.org
>     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
>
>     =20
>     I had typed this response out earlier but wanted to see if there
>     was any support for this capability. So sending it out now.
>     =20
>     The solution I would suggest is a new header
>                Type-Ahead: <time-in-milli-seconds>
>     header that can be sent on the SET-PARAMS, GET-PARAMS and
>     RECOGNIZE methods.
>     =20
>     default is 0 which is todays no buffering case. If set to some
>     value, the resource is expected to buffer upto
>     <time-in-milliseconds> of audio and use it during the recognize as
>     a type-ahead buffer. I can see this being a boolean field as well,
>     but the <time-in-milliseconds> approach is little more flexible.
>     =20
>     Sarvi
>
>
------------------------------------------------------------------------
>         *From:* Reifenrath, Klaus
[mailto:Klaus.Reifenrath@Scansoft.com]
>         *Sent:* Tuesday, August 23, 2005 8:41 AM
>         *To:* Shanmugham, Saravanan; 'dburnett@vocalocity.net'
>         *Cc:* 'speechsc@ietf.org'
>         *Subject:* Problem with DTMF type-ahead
>
>         The MRCP and MRCPv2 specifications define how DTMF input
>         should be handled: DTMF input is collected after the RECOGNIZE
>         request is received until a term-char or timeout is reached.
>         Then a result or no-match is returned based on any active DTMF
>         grammars. The problem is there can be a significant amount of
>         time between when one result is processed by the application
>         and the next grammar is activated. During that time any DTMF
>         input received will be lost. Such DTMF type-ahead is common
>         for frequent users of IVR systems, and buffering that type
>         ahead is required by VoiceXML 2.0 (see section 4.1.8). Can
>         this requirement of VoiceXML be implemented using MRCP
>         (assuming that the media source CANNOT buffer DTMF)? If so,
>         how is it done?
>
>         Thanks,
>         Klaus
>
>-----------------------------------------------------------------------
-
>
>_______________________________________________
>Speechsc mailing list
>Speechsc@ietf.org
>https://www1.ietf.org/mailman/listinfo/speechsc
> =20
>

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

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



From speechsc-bounces@ietf.org Wed Aug 24 02:47:29 2005
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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
Date: Tue, 23 Aug 2005 23:43:38 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C4253@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead
Thread-Index: AcWoO0FVkK5nyeW4Sn2qclBDDIBrlwANxhggAAD+NRA=
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Eric Burger" <eburger@brooktrout.com>, <speechsc@ietf.org>
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The idea of being able to send it opn the RECOGNIZE is to reduce the
time and not increase it.=20

By default, the value is 0. When you do a SET-PARAMS and set it to say
10000 it sets the buffering at the server to 10 seconds.

This does not mean that every RECOGNIZE should use this entire 10s of
buffer. On an individual RECOGNIZE method basis the client should be
able to use 0 - 10s of that buffer. This would be the last X seconds of
the buffer before the RECOGNIZE method is recieved the rest would be
discarded.

Achieves more flexibility than plain flushing true or false.

Sarvi

     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Eric Burger
     Sent: Tuesday, August 23, 2005 11:14 PM
     To: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     [not as chair]
    =20
     I can see something like "Type-Ahead" for SET/GET-PARAMS,=20
     which sets a global digit buffer.  I would suggest a more=20
     descriptive name, like DTMF-Buffer-Time.
    =20
     That said, I don't see it working for the RECOGNIZE=20
     method.  The speech resource needs to know *before* the=20
     RECOGNIZE request that it needs to buffer DTMF.  If it=20
     comes in with the RECOGNIZE request, it will be too late.
    =20
     For example:
    =20
     SET-PARAMS
     DTMF-Buffer-Time: 10000
     ...
     <sets global buffer to 10s; platform keeps 10 seconds of=20
     DTMF buffered>
    =20
    =20
     RECOGNIZE
     DTMF-Buffer-Time: 20000
     ...
     <sets buffer to 20s.  Oops: platform already dropped first=20
     10 seconds because it was only buffering 10 seconds>
    =20
    =20
    =20
     The sensible per-RECOGNIZE attribute is to flush the=20
     buffer.  While one might be tempted to say=20
     "DTMF-Buffer-Time: 0" means flush the buffer, I am sure=20
     someone will try to use it with a non-zero value.  I would=20
     offer we use a different attribute to indicate flushing=20
     the buffer before recognition, like "Flush-DTMF: true".
    =20
     -----Original Message-----
     From: speechsc-bounces@ietf.org=20
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Corby Anderson
     Sent: Tuesday, August 23, 2005 7:34 PM
     To: speechsc@ietf.org
     Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     I prefer the flexible "Type-Ahead: <time-in-milliseconds>"=20
     approach. =20
     This would be useful for implementing vxml listen states=20
     that happen after data tags.  If a data fetch takes longer=20
     than several seconds, you
    =20
     might not want to play dtmf that is older than some limit.
    =20
     Corby Anderson
     Tellme Networks, Inc.
    =20
     Shanmugham, Saravanan wrote:
    =20
     > Are we assuming then that this is only limited to DTMF=20
     type-ahead and=20
     > is not speak ahead as well?
     > =20
     > Sarvi
     >
     >
     -----------------------------------------------------------
     -------------
     >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
     >     *Sent:* Tuesday, August 23, 2005 3:20 PM
     >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
     >     dburnett@vocalocity.net
     >     *Cc:* speechsc@ietf.org
     >     *Subject:* RE: [Speechsc] RE: Problem with DTMF type-ahead
     >
     >     It would seem like we're not trying to buffer audio=20
     (generally)
     >     but only the DIGIT values themselves.=20
     >
     >     It would seem better to just be a header such as:
     >     =20
     >     DTMF-Buffer: <true|false>
     >     =20
     >     instead.
     >     =20
     >     Jeff
     >
     >
     -----------------------------------------------------------
     -------------
     >     *From:* speechsc-bounces@ietf.org
     >     [mailto:speechsc-bounces@ietf.org] *On Behalf Of *Shanmugham,
     >     Saravanan
     >     *Sent:* Tuesday, August 23, 2005 4:59 PM
     >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
     >     *Cc:* speechsc@ietf.org
     >     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
     >
     >     =20
     >     I had typed this response out earlier but wanted to=20
     see if there
     >     was any support for this capability. So sending it out now.
     >     =20
     >     The solution I would suggest is a new header
     >                Type-Ahead: <time-in-milli-seconds>
     >     header that can be sent on the SET-PARAMS, GET-PARAMS and
     >     RECOGNIZE methods.
     >     =20
     >     default is 0 which is todays no buffering case. If=20
     set to some
     >     value, the resource is expected to buffer upto
     >     <time-in-milliseconds> of audio and use it during=20
     the recognize as
     >     a type-ahead buffer. I can see this being a boolean=20
     field as well,
     >     but the <time-in-milliseconds> approach is little=20
     more flexible.
     >     =20
     >     Sarvi
     >
     >
     -----------------------------------------------------------
     -------------
     >         *From:* Reifenrath, Klaus
     [mailto:Klaus.Reifenrath@Scansoft.com]
     >         *Sent:* Tuesday, August 23, 2005 8:41 AM
     >         *To:* Shanmugham, Saravanan; 'dburnett@vocalocity.net'
     >         *Cc:* 'speechsc@ietf.org'
     >         *Subject:* Problem with DTMF type-ahead
     >
     >         The MRCP and MRCPv2 specifications define how DTMF input
     >         should be handled: DTMF input is collected after=20
     the RECOGNIZE
     >         request is received until a term-char or timeout=20
     is reached.
     >         Then a result or no-match is returned based on=20
     any active DTMF
     >         grammars. The problem is there can be a=20
     significant amount of
     >         time between when one result is processed by the=20
     application
     >         and the next grammar is activated. During that=20
     time any DTMF
     >         input received will be lost. Such DTMF=20
     type-ahead is common
     >         for frequent users of IVR systems, and buffering=20
     that type
     >         ahead is required by VoiceXML 2.0 (see section=20
     4.1.8). Can
     >         this requirement of VoiceXML be implemented using MRCP
     >         (assuming that the media source CANNOT buffer=20
     DTMF)? If so,
     >         how is it done?
     >
     >         Thanks,
     >         Klaus
     >
     >----------------------------------------------------------
     -------------
     -
     >
     >_______________________________________________
     >Speechsc mailing list
     >Speechsc@ietf.org
     >https://www1.ietf.org/mailman/listinfo/speechsc
     > =20
     >
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

_______________________________________________
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From speechsc-bounces@ietf.org Wed Aug 24 05:52:48 2005
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From: "Dave Burke" <david.burke@voxpilot.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Eric Burger" <eburger@brooktrout.com>, <speechsc@ietf.org>
References: <03772D1EC8DE624A863058C75874A75C2C4253@vtg-um-e2k6.sj21ad.cisco.com>
Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
Date: Wed, 24 Aug 2005 10:52:20 +0100
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Sounds like a lot of dials to play with: What is the use-case warranting the 
extra complexity for overriding the SET-PARAMed buffer length on a per 
RECOGNIZE?

The two capabilities that you want (IMO) are:
    1. Configuration: How long is the buffer?
    2. Action: Flush buffer

For configuration, I'd rather see a SET-PARAMs but with a non-zero default - 
say 10 seconds (2 x default inter-digit time).  For flushing, I'd rather see 
a boolean header on RECOGNIZE for simplicity. That way, out-of-the-box, easy 
things are easy and harder things are still possible...

I don't like the term flush. Is one flushing the DTMF into the recognizer or 
out into the ether(net)? Hence why I suggested Clear-DTMF-Buffer.

Dave

----- Original Message ----- 
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Eric Burger" <eburger@brooktrout.com>; <speechsc@ietf.org>
Sent: Wednesday, August 24, 2005 7:43 AM
Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead


The idea of being able to send it opn the RECOGNIZE is to reduce the
time and not increase it.

By default, the value is 0. When you do a SET-PARAMS and set it to say
10000 it sets the buffering at the server to 10 seconds.

This does not mean that every RECOGNIZE should use this entire 10s of
buffer. On an individual RECOGNIZE method basis the client should be
able to use 0 - 10s of that buffer. This would be the last X seconds of
the buffer before the RECOGNIZE method is recieved the rest would be
discarded.

Achieves more flexibility than plain flushing true or false.

Sarvi

     -----Original Message-----
     From: speechsc-bounces@ietf.org
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Eric Burger
     Sent: Tuesday, August 23, 2005 11:14 PM
     To: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead

     [not as chair]

     I can see something like "Type-Ahead" for SET/GET-PARAMS,
     which sets a global digit buffer.  I would suggest a more
     descriptive name, like DTMF-Buffer-Time.

     That said, I don't see it working for the RECOGNIZE
     method.  The speech resource needs to know *before* the
     RECOGNIZE request that it needs to buffer DTMF.  If it
     comes in with the RECOGNIZE request, it will be too late.

     For example:

     SET-PARAMS
     DTMF-Buffer-Time: 10000
     ...
     <sets global buffer to 10s; platform keeps 10 seconds of
     DTMF buffered>


     RECOGNIZE
     DTMF-Buffer-Time: 20000
     ...
     <sets buffer to 20s.  Oops: platform already dropped first
     10 seconds because it was only buffering 10 seconds>



     The sensible per-RECOGNIZE attribute is to flush the
     buffer.  While one might be tempted to say
     "DTMF-Buffer-Time: 0" means flush the buffer, I am sure
     someone will try to use it with a non-zero value.  I would
     offer we use a different attribute to indicate flushing
     the buffer before recognition, like "Flush-DTMF: true".

     -----Original Message-----
     From: speechsc-bounces@ietf.org
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Corby Anderson
     Sent: Tuesday, August 23, 2005 7:34 PM
     To: speechsc@ietf.org
     Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead

     I prefer the flexible "Type-Ahead: <time-in-milliseconds>"
     approach.
     This would be useful for implementing vxml listen states
     that happen after data tags.  If a data fetch takes longer
     than several seconds, you

     might not want to play dtmf that is older than some limit.

     Corby Anderson
     Tellme Networks, Inc.

     Shanmugham, Saravanan wrote:

     > Are we assuming then that this is only limited to DTMF
     type-ahead and
     > is not speak ahead as well?
     >
     > Sarvi
     >
     >
     -----------------------------------------------------------
     -------------
     >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
     >     *Sent:* Tuesday, August 23, 2005 3:20 PM
     >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
     >     dburnett@vocalocity.net
     >     *Cc:* speechsc@ietf.org
     >     *Subject:* RE: [Speechsc] RE: Problem with DTMF type-ahead
     >
     >     It would seem like we're not trying to buffer audio
     (generally)
     >     but only the DIGIT values themselves.
     >
     >     It would seem better to just be a header such as:
     >
     >     DTMF-Buffer: <true|false>
     >
     >     instead.
     >
     >     Jeff
     >
     >
     -----------------------------------------------------------
     -------------
     >     *From:* speechsc-bounces@ietf.org
     >     [mailto:speechsc-bounces@ietf.org] *On Behalf Of *Shanmugham,
     >     Saravanan
     >     *Sent:* Tuesday, August 23, 2005 4:59 PM
     >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
     >     *Cc:* speechsc@ietf.org
     >     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
     >
     >
     >     I had typed this response out earlier but wanted to
     see if there
     >     was any support for this capability. So sending it out now.
     >
     >     The solution I would suggest is a new header
     >                Type-Ahead: <time-in-milli-seconds>
     >     header that can be sent on the SET-PARAMS, GET-PARAMS and
     >     RECOGNIZE methods.
     >
     >     default is 0 which is todays no buffering case. If
     set to some
     >     value, the resource is expected to buffer upto
     >     <time-in-milliseconds> of audio and use it during
     the recognize as
     >     a type-ahead buffer. I can see this being a boolean
     field as well,
     >     but the <time-in-milliseconds> approach is little
     more flexible.
     >
     >     Sarvi
     >
     >
     -----------------------------------------------------------
     -------------
     >         *From:* Reifenrath, Klaus
     [mailto:Klaus.Reifenrath@Scansoft.com]
     >         *Sent:* Tuesday, August 23, 2005 8:41 AM
     >         *To:* Shanmugham, Saravanan; 'dburnett@vocalocity.net'
     >         *Cc:* 'speechsc@ietf.org'
     >         *Subject:* Problem with DTMF type-ahead
     >
     >         The MRCP and MRCPv2 specifications define how DTMF input
     >         should be handled: DTMF input is collected after
     the RECOGNIZE
     >         request is received until a term-char or timeout
     is reached.
     >         Then a result or no-match is returned based on
     any active DTMF
     >         grammars. The problem is there can be a
     significant amount of
     >         time between when one result is processed by the
     application
     >         and the next grammar is activated. During that
     time any DTMF
     >         input received will be lost. Such DTMF
     type-ahead is common
     >         for frequent users of IVR systems, and buffering
     that type
     >         ahead is required by VoiceXML 2.0 (see section
     4.1.8). Can
     >         this requirement of VoiceXML be implemented using MRCP
     >         (assuming that the media source CANNOT buffer
     DTMF)? If so,
     >         how is it done?
     >
     >         Thanks,
     >         Klaus
     >
     >----------------------------------------------------------
     -------------
     -
     >
     >_______________________________________________
     >Speechsc mailing list
     >Speechsc@ietf.org
     >https://www1.ietf.org/mailman/listinfo/speechsc
     >
     >

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

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From speechsc-bounces@ietf.org Wed Aug 24 07:36:34 2005
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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
Date: Wed, 24 Aug 2005 07:31:13 -0400
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Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead
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From: "Eric Burger" <eburger@brooktrout.com>
To: "Dave Burke" <david.burke@voxpilot.com>,
	"Shanmugham, Saravanan" <sarvi@cisco.com>
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[not as chair]
We went through this whole discussion for KPML.  No one ever had a use
case for "keep the last x seconds".  In fact, one either buffered digits
or flushed them.

If you have an 8,000 port media engine, with one hour call times, and
continuous digit entry without the digits ever getting off the queue,
and the digit entry is computer generated (i.e., as fast as possible),
you need:
   (8,000 ports) * (3,600,000ms/port) / (60ms / digit) =3D 480MB

I would offer that if you have an 8,000 port media processing device,
less than half a gigabyte for digit buffering is nothing.

In English, that means that you can realistically have infinite digit
buffering.

That makes the protocol MUCH easier, interoperation MUCH easier, and
implementation MUCH easier.

[really not as chair]
As for the term, I didn't think anyone had an alternative interpretation
of the term "flush".  However, I suppose "clear" is OK.

-----Original Message-----
From: Dave Burke [mailto:david.burke@voxpilot.com]=20
Sent: Wednesday, August 24, 2005 5:52 AM
To: Shanmugham, Saravanan; Eric Burger; speechsc@ietf.org
Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead

Sounds like a lot of dials to play with: What is the use-case warranting
the=20
extra complexity for overriding the SET-PARAMed buffer length on a per=20
RECOGNIZE?

The two capabilities that you want (IMO) are:
    1. Configuration: How long is the buffer?
    2. Action: Flush buffer

For configuration, I'd rather see a SET-PARAMs but with a non-zero
default -=20
say 10 seconds (2 x default inter-digit time).  For flushing, I'd rather
see=20
a boolean header on RECOGNIZE for simplicity. That way, out-of-the-box,
easy=20
things are easy and harder things are still possible...

I don't like the term flush. Is one flushing the DTMF into the
recognizer or=20
out into the ether(net)? Hence why I suggested Clear-DTMF-Buffer.

Dave

----- Original Message -----=20
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Eric Burger" <eburger@brooktrout.com>; <speechsc@ietf.org>
Sent: Wednesday, August 24, 2005 7:43 AM
Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead


The idea of being able to send it opn the RECOGNIZE is to reduce the
time and not increase it.

By default, the value is 0. When you do a SET-PARAMS and set it to say
10000 it sets the buffering at the server to 10 seconds.

This does not mean that every RECOGNIZE should use this entire 10s of
buffer. On an individual RECOGNIZE method basis the client should be
able to use 0 - 10s of that buffer. This would be the last X seconds of
the buffer before the RECOGNIZE method is recieved the rest would be
discarded.

Achieves more flexibility than plain flushing true or false.

Sarvi

     -----Original Message-----
     From: speechsc-bounces@ietf.org
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Eric Burger
     Sent: Tuesday, August 23, 2005 11:14 PM
     To: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead

     [not as chair]

     I can see something like "Type-Ahead" for SET/GET-PARAMS,
     which sets a global digit buffer.  I would suggest a more
     descriptive name, like DTMF-Buffer-Time.

     That said, I don't see it working for the RECOGNIZE
     method.  The speech resource needs to know *before* the
     RECOGNIZE request that it needs to buffer DTMF.  If it
     comes in with the RECOGNIZE request, it will be too late.

     For example:

     SET-PARAMS
     DTMF-Buffer-Time: 10000
     ...
     <sets global buffer to 10s; platform keeps 10 seconds of
     DTMF buffered>


     RECOGNIZE
     DTMF-Buffer-Time: 20000
     ...
     <sets buffer to 20s.  Oops: platform already dropped first
     10 seconds because it was only buffering 10 seconds>



     The sensible per-RECOGNIZE attribute is to flush the
     buffer.  While one might be tempted to say
     "DTMF-Buffer-Time: 0" means flush the buffer, I am sure
     someone will try to use it with a non-zero value.  I would
     offer we use a different attribute to indicate flushing
     the buffer before recognition, like "Flush-DTMF: true".

     -----Original Message-----
     From: speechsc-bounces@ietf.org
     [mailto:speechsc-bounces@ietf.org] On Behalf Of Corby Anderson
     Sent: Tuesday, August 23, 2005 7:34 PM
     To: speechsc@ietf.org
     Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead

     I prefer the flexible "Type-Ahead: <time-in-milliseconds>"
     approach.
     This would be useful for implementing vxml listen states
     that happen after data tags.  If a data fetch takes longer
     than several seconds, you

     might not want to play dtmf that is older than some limit.

     Corby Anderson
     Tellme Networks, Inc.

     Shanmugham, Saravanan wrote:

     > Are we assuming then that this is only limited to DTMF
     type-ahead and
     > is not speak ahead as well?
     >
     > Sarvi
     >
     >
     -----------------------------------------------------------
     -------------
     >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
     >     *Sent:* Tuesday, August 23, 2005 3:20 PM
     >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
     >     dburnett@vocalocity.net
     >     *Cc:* speechsc@ietf.org
     >     *Subject:* RE: [Speechsc] RE: Problem with DTMF type-ahead
     >
     >     It would seem like we're not trying to buffer audio
     (generally)
     >     but only the DIGIT values themselves.
     >
     >     It would seem better to just be a header such as:
     >
     >     DTMF-Buffer: <true|false>
     >
     >     instead.
     >
     >     Jeff
     >
     >
     -----------------------------------------------------------
     -------------
     >     *From:* speechsc-bounces@ietf.org
     >     [mailto:speechsc-bounces@ietf.org] *On Behalf Of *Shanmugham,
     >     Saravanan
     >     *Sent:* Tuesday, August 23, 2005 4:59 PM
     >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
     >     *Cc:* speechsc@ietf.org
     >     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
     >
     >
     >     I had typed this response out earlier but wanted to
     see if there
     >     was any support for this capability. So sending it out now.
     >
     >     The solution I would suggest is a new header
     >                Type-Ahead: <time-in-milli-seconds>
     >     header that can be sent on the SET-PARAMS, GET-PARAMS and
     >     RECOGNIZE methods.
     >
     >     default is 0 which is todays no buffering case. If
     set to some
     >     value, the resource is expected to buffer upto
     >     <time-in-milliseconds> of audio and use it during
     the recognize as
     >     a type-ahead buffer. I can see this being a boolean
     field as well,
     >     but the <time-in-milliseconds> approach is little
     more flexible.
     >
     >     Sarvi
     >
     >
     -----------------------------------------------------------
     -------------
     >         *From:* Reifenrath, Klaus
     [mailto:Klaus.Reifenrath@Scansoft.com]
     >         *Sent:* Tuesday, August 23, 2005 8:41 AM
     >         *To:* Shanmugham, Saravanan; 'dburnett@vocalocity.net'
     >         *Cc:* 'speechsc@ietf.org'
     >         *Subject:* Problem with DTMF type-ahead
     >
     >         The MRCP and MRCPv2 specifications define how DTMF input
     >         should be handled: DTMF input is collected after
     the RECOGNIZE
     >         request is received until a term-char or timeout
     is reached.
     >         Then a result or no-match is returned based on
     any active DTMF
     >         grammars. The problem is there can be a
     significant amount of
     >         time between when one result is processed by the
     application
     >         and the next grammar is activated. During that
     time any DTMF
     >         input received will be lost. Such DTMF
     type-ahead is common
     >         for frequent users of IVR systems, and buffering
     that type
     >         ahead is required by VoiceXML 2.0 (see section
     4.1.8). Can
     >         this requirement of VoiceXML be implemented using MRCP
     >         (assuming that the media source CANNOT buffer
     DTMF)? If so,
     >         how is it done?
     >
     >         Thanks,
     >         Klaus
     >
     >----------------------------------------------------------
     -------------
     -
     >
     >_______________________________________________
     >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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     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc


_______________________________________________
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_______________________________________________
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To: "Shanmugham, Saravanan" <sarvi@cisco.com>
Subject: Re: [Speechsc] Recognition Completion-Cause Codes 003 and 008
	revisited
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The problem with re-using one of the redundant completion-cause codes is 
that we changed the meaning of the other one -- it has been limited to 
cases where we have a match. Now neither of the completion-cause codes 
map to VoiceXML maxspeechtimeout -- there was a recognition-timeout and 
there was no match. I think we still need a code for this case.

The explanation for the hotword completion-cause code doesn't make sense 
or I'm missing something. You're saying that the recognition would still 
continue when the timer fires even when there is no match. This makes 
sense for hotword recognition, but if recognition continues, then why do 
you need a completion-cause code... recognition has not completed.

Actually, now that I think about it, in the context of hotword 
recognition, the recognition-timeout seems to be identical to the 
hotword-max-duration. Is this accurate?

Andrew Wahbe
VoiceGenie Technologies

Shanmugham, Saravanan wrote:

>I was going to remove Recognition-Timeout based on the thread on the
>confusion between this cause and max-speech-timeout etc.
>
>But it was agreed we need a timeout capability for HotWord Recognition
>based ona separate thread.
>
>In the hotword case, none of the timeout desciptions match the way
>timeout happens in hotword recognition. Since there, even if the words
>spoken until that point did not match any of the hotwords, the
>recognition would still continue. Hence a separate completion code for
>hotword recognition timeout was needed. And I have used 03 for that use
>case.
>
>Sarvi
>
>     -----Original Message-----
>     From: speechsc-bounces@ietf.org 
>     [mailto:speechsc-bounces@ietf.org] On Behalf Of Andrew Wahbe
>     Sent: Tuesday, August 23, 2005 7:34 AM
>     To: speechsc@ietf.org
>     Subject: [Speechsc] Recognition Completion-Cause Codes 003 
>     and 008 revisited
>     
>     A few weeks ago, there was an extensive discussion about 
>     the recognition completion-cause codes for cases where 
>     recognition-timers fire. There seemed to be two codes, 003 
>     and 008, that meant the same thing -- the recognition was 
>     aborted due to the recognition-timer firing. This case 
>     maps to the maxspeechtimeout event in VoiceXML.
>     
>     A lot of hypothesizing was done to try and figure out what 
>     the difference between the codes was. After some 
>     discussion it seems we ended up with no codes that map to 
>     maxspeechtimeout in VoiceXML (i.e. 
>     the recognition was aborted because of recognition-timeout 
>     and there is no match).
>     
>     I think things would be set right if 003 was changed from:
>     
>        | 003    | recognition-timeout   | RECOGNIZE in hotword 
>     mode        |
>        |        |                       | completed without a 
>     match due to |
>        |        |                       | a 
>     recognition-timeout            |
>     
>     to:
>     
>        | 003    | recognition-timeout   | RECOGNIZE completed 
>     without a    |
>        |        |                       | match due to a       
>                 |
>        |        |                       | recognition-timeout  
>                 |
>     
>     
>     I am not sure why a special code is needed for this timer 
>     firing in hotword mode. Am I missing something?
>     
>     Andrew Wahbe
>     VoiceGenie Technologies
>     
>
>
>  
>

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--------------000607080108070700050700--




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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead 
Date: Wed, 24 Aug 2005 14:01:01 -0400
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Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead 
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From: "Pierre Forgues" <forgues@nuance.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>,
	"Jeff Haynie" <jhaynie@vocalocity.net>,
	"Klaus Reifenrath" <Klaus.Reifenrath@Scansoft.com>,
	<dburnett@vocalocity.net>
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Let's limit the capability to DTMF type-ahead.  I've heard of power
users, but I don't think it is feasible or desirable to talk about
people speaking before the beginning of prompts.  This would add
unnecessary complexity in my opinion.

=20

I like the terminology of "clearing" a buffer rather than flushing.

=20

Pierre

=20

________________________________

From: speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] On
Behalf Of Shanmugham, Saravanan
Sent: Tuesday, August 23, 2005 6:31 PM
To: Jeff Haynie; Klaus Reifenrath; dburnett@vocalocity.net
Cc: speechsc@ietf.org
Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead=20

=20

Are we assuming then that this is only limited to DTMF type-ahead and is
not speak ahead as well?

=20

Sarvi

	=20

=09
________________________________


	From: Jeff Haynie [mailto:jhaynie@vocalocity.net]=20
	Sent: Tuesday, August 23, 2005 3:20 PM
	To: Shanmugham, Saravanan; Reifenrath, Klaus;
dburnett@vocalocity.net
	Cc: speechsc@ietf.org
	Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead=20

	It would seem like we're not trying to buffer audio (generally)
but only the DIGIT values themselves. =20

=09
	It would seem better to just be a header such as:

	=20

	DTMF-Buffer: <true|false>

	=20

	instead.

	=20

	Jeff

	=20

=09
________________________________


	From: speechsc-bounces@ietf.org
[mailto:speechsc-bounces@ietf.org] On Behalf Of Shanmugham, Saravanan
	Sent: Tuesday, August 23, 2005 4:59 PM
	To: Reifenrath, Klaus; dburnett@vocalocity.net
	Cc: speechsc@ietf.org
	Subject: [Speechsc] RE: Problem with DTMF type-ahead=20

	=20

	I had typed this response out earlier but wanted to see if there
was any support for this capability. So sending it out now.

	=20

	The solution I would suggest is a new header=20

	           Type-Ahead: <time-in-milli-seconds>=20

	header that can be sent on the SET-PARAMS, GET-PARAMS and
RECOGNIZE methods.

	=20

	default is 0 which is todays no buffering case. If set to some
value, the resource is expected to buffer upto <time-in-milliseconds> of
audio and use it during the recognize as a type-ahead buffer. I can see
this being a boolean field as well, but the <time-in-milliseconds>
approach is little more flexible.

	=20

	Sarvi

		=20

	=09
________________________________


		From: Reifenrath, Klaus
[mailto:Klaus.Reifenrath@Scansoft.com]=20
		Sent: Tuesday, August 23, 2005 8:41 AM
		To: Shanmugham, Saravanan; 'dburnett@vocalocity.net'
		Cc: 'speechsc@ietf.org'
		Subject: Problem with DTMF type-ahead=20

		The MRCP and MRCPv2 specifications define how DTMF input
should be handled: DTMF input is collected after the RECOGNIZE request
is received until a term-char or timeout is reached. Then a result or
no-match is returned based on any active DTMF grammars. The problem is
there can be a significant amount of time between when one result is
processed by the application and the next grammar is activated. During
that time any DTMF input received will be lost. Such DTMF type-ahead is
common for frequent users of IVR systems, and buffering that type ahead
is required by VoiceXML 2.0 (see section 4.1.8). Can this requirement of
VoiceXML be implemented using MRCP (assuming that the media source
CANNOT buffer DTMF)? If so, how is it done?=20

		Thanks,
		Klaus=20


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Let&#8217;s limit the capability to =
DTMF
type-ahead. &nbsp;I&#8217;ve heard of power users, but I don&#8217;t =
think it
is feasible or desirable to talk about people speaking before the =
beginning of
prompts. &nbsp;This would add unnecessary complexity in my =
opinion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I like the terminology of =
&#8220;clearing&#8221;
a buffer rather than flushing.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><st1:City w:st=3D"on"><st1:place w:st=3D"on"><font =
size=3D2
  color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
  color:navy'>Pierre</span></font></st1:place></st1:City><font size=3D2
color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

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face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

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</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Shanmugham, =
Saravanan<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, August 23, =
2005
6:31 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jeff Haynie; =
<st1:PersonName
w:st=3D"on">Klaus Reifenrath</st1:PersonName>; =
dburnett@vocalocity.net<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> speechsc@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Speechsc] =
RE:
Problem with DTMF type-ahead </span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Are we assuming then that this is
only&nbsp;limited to&nbsp;DTMF type-ahead and is not speak ahead as =
well?</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Sarvi</span></font><o:p></o:p></p>

<blockquote style=3D'border:none;border-left:solid blue =
1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
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<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
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</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Jeff
Haynie [mailto:jhaynie@vocalocity.net] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, August 23, =
2005
3:20 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Shanmugham, =
Saravanan;
Reifenrath, Klaus; dburnett@vocalocity.net<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> speechsc@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Speechsc] =
RE:
Problem with DTMF type-ahead </span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>It would seem like we're not trying =
to
buffer audio (generally) but only the DIGIT values themselves.&nbsp; =
</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><br>
</span></font><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;color:blue'>It would seem better to just be a header =
such as:</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>DTMF-Buffer: =
&lt;true|false&gt;</span></font><o:p></o:p></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>instead.</span></font><o:p></o:p></p=
>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Jeff</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>
speechsc-bounces@ietf.org [mailto:speechsc-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Shanmugham, =
Saravanan<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, August 23, =
2005
4:59 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Reifenrath, Klaus;
dburnett@vocalocity.net<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> speechsc@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Speechsc] RE: =
Problem
with DTMF type-ahead </span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>I had typed this response out =
earlier but
wanted to see if there was any support for this capability. So sending =
it out
now.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>The solution I would suggest is a =
new
header </span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;
Type-Ahead: &lt;time-in-milli-seconds&gt; </span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>header that can be sent on the =
SET-PARAMS,
GET-PARAMS and RECOGNIZE methods.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>default is 0 which is todays no =
buffering
case. If set to some value, the resource is expected to buffer upto
&lt;time-in-milliseconds&gt; of audio and use it during the recognize as =
a type-ahead
buffer. I can see this being a boolean field as well, but the
&lt;time-in-milliseconds&gt; approach is little more =
flexible.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Sarvi</span></font><o:p></o:p></p>

<blockquote style=3D'border:none;border-left:solid blue =
1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Reifenrath, Klaus [mailto:Klaus.Reifenrath@Scansoft.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, August 23, =
2005
8:41 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Shanmugham, =
Saravanan;
'dburnett@vocalocity.net'<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> =
'speechsc@ietf.org'<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Problem with =
DTMF
type-ahead </span></font><o:p></o:p></p>

<p><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>The
MRCP and MRCPv2 specifications define how DTMF input should be handled: =
DTMF
input is collected after the RECOGNIZE request is received until a =
term-char or
timeout is reached. Then a result or no-match is returned based on any =
active
DTMF grammars. The problem is there can be a significant amount of time =
between
when one result is processed by the application and the next grammar is
activated. During that time any DTMF input received will be lost. Such =
DTMF
type-ahead is common for frequent users of IVR systems, and buffering =
that type
ahead is required by VoiceXML 2.0 (see section 4.1.8).</span></font> =
Can<font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> this
requirement of VoiceXML be implemented using MRCP (assuming that the =
media
source CANNOT buffer DTMF)? If so, how is it done? =
</span></font><o:p></o:p></p>

<p><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Thanks,<br>
Klaus</span></font> <o:p></o:p></p>

</blockquote>

</blockquote>

</div>

</body>

</html>

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From speechsc-bounces@ietf.org Wed Aug 24 14:22:31 2005
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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
Date: Wed, 24 Aug 2005 11:22:10 -0700
Message-ID: <03772D1EC8DE624A863058C75874A75C2C42A9@vtg-um-e2k6.sj21ad.cisco.com>
Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead
Thread-Index: AcWokY2nOWUMZ0/RTlajUyiwMSDepAADFCrAAA5tVzA=
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Eric Burger" <eburger@brooktrout.com>,
	"Dave Burke" <david.burke@voxpilot.com>
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Ok.
Lets limit this buffer to digits and not speech.=20
Lets also stick to clear buffer operation in RECOGNIZE method.

But should be the buffer size. Settable by the client? Platform specific
fixed? Settable client on a per RECOGNIZE basis.

Unlimited size may be ok for implementation, but is it practical and
what does it mean in actual usage.=20
If the recognizer channel was allocated at the beginning of the call,
and at some later point in the call, much later, we do a recognize. Does
this mean that any digits pressed since the beginning of the call when
the resource was allocated or the previous RECOGNIZE, are used as part
of the type-ahead?

Sarvi=20

     -----Original Message-----
     From: Eric Burger [mailto:eburger@brooktrout.com]=20
     Sent: Wednesday, August 24, 2005 4:31 AM
     To: Dave Burke; Shanmugham, Saravanan
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     [not as chair]
     We went through this whole discussion for KPML.  No one=20
     ever had a use case for "keep the last x seconds".  In=20
     fact, one either buffered digits or flushed them.
    =20
     If you have an 8,000 port media engine, with one hour call=20
     times, and continuous digit entry without the digits ever=20
     getting off the queue, and the digit entry is computer=20
     generated (i.e., as fast as possible), you need:
        (8,000 ports) * (3,600,000ms/port) / (60ms / digit) =3D 480MB
    =20
     I would offer that if you have an 8,000 port media=20
     processing device, less than half a gigabyte for digit=20
     buffering is nothing.
    =20
     In English, that means that you can realistically have=20
     infinite digit buffering.
    =20
     That makes the protocol MUCH easier, interoperation MUCH=20
     easier, and implementation MUCH easier.
    =20
     [really not as chair]
     As for the term, I didn't think anyone had an alternative=20
     interpretation of the term "flush".  However, I suppose=20
     "clear" is OK.
    =20
     -----Original Message-----
     From: Dave Burke [mailto:david.burke@voxpilot.com]
     Sent: Wednesday, August 24, 2005 5:52 AM
     To: Shanmugham, Saravanan; Eric Burger; speechsc@ietf.org
     Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     Sounds like a lot of dials to play with: What is the=20
     use-case warranting the extra complexity for overriding=20
     the SET-PARAMed buffer length on a per RECOGNIZE?
    =20
     The two capabilities that you want (IMO) are:
         1. Configuration: How long is the buffer?
         2. Action: Flush buffer
    =20
     For configuration, I'd rather see a SET-PARAMs but with a=20
     non-zero default - say 10 seconds (2 x default inter-digit=20
     time).  For flushing, I'd rather see a boolean header on=20
     RECOGNIZE for simplicity. That way, out-of-the-box, easy=20
     things are easy and harder things are still possible...
    =20
     I don't like the term flush. Is one flushing the DTMF into=20
     the recognizer or out into the ether(net)? Hence why I=20
     suggested Clear-DTMF-Buffer.
    =20
     Dave
    =20
     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Eric Burger" <eburger@brooktrout.com>; <speechsc@ietf.org>
     Sent: Wednesday, August 24, 2005 7:43 AM
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
    =20
     The idea of being able to send it opn the RECOGNIZE is to=20
     reduce the
     time and not increase it.
    =20
     By default, the value is 0. When you do a SET-PARAMS and=20
     set it to say
     10000 it sets the buffering at the server to 10 seconds.
    =20
     This does not mean that every RECOGNIZE should use this=20
     entire 10s of
     buffer. On an individual RECOGNIZE method basis the client=20
     should be
     able to use 0 - 10s of that buffer. This would be the last=20
     X seconds of
     the buffer before the RECOGNIZE method is recieved the=20
     rest would be
     discarded.
    =20
     Achieves more flexibility than plain flushing true or false.
    =20
     Sarvi
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Eric Burger
          Sent: Tuesday, August 23, 2005 11:14 PM
          To: speechsc@ietf.org
          Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
          [not as chair]
    =20
          I can see something like "Type-Ahead" for SET/GET-PARAMS,
          which sets a global digit buffer.  I would suggest a more
          descriptive name, like DTMF-Buffer-Time.
    =20
          That said, I don't see it working for the RECOGNIZE
          method.  The speech resource needs to know *before* the
          RECOGNIZE request that it needs to buffer DTMF.  If it
          comes in with the RECOGNIZE request, it will be too late.
    =20
          For example:
    =20
          SET-PARAMS
          DTMF-Buffer-Time: 10000
          ...
          <sets global buffer to 10s; platform keeps 10 seconds of
          DTMF buffered>
    =20
    =20
          RECOGNIZE
          DTMF-Buffer-Time: 20000
          ...
          <sets buffer to 20s.  Oops: platform already dropped first
          10 seconds because it was only buffering 10 seconds>
    =20
    =20
    =20
          The sensible per-RECOGNIZE attribute is to flush the
          buffer.  While one might be tempted to say
          "DTMF-Buffer-Time: 0" means flush the buffer, I am sure
          someone will try to use it with a non-zero value.  I would
          offer we use a different attribute to indicate flushing
          the buffer before recognition, like "Flush-DTMF: true".
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Corby Anderson
          Sent: Tuesday, August 23, 2005 7:34 PM
          To: speechsc@ietf.org
          Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
    =20
          I prefer the flexible "Type-Ahead: <time-in-milliseconds>"
          approach.
          This would be useful for implementing vxml listen states
          that happen after data tags.  If a data fetch takes longer
          than several seconds, you
    =20
          might not want to play dtmf that is older than some limit.
    =20
          Corby Anderson
          Tellme Networks, Inc.
    =20
          Shanmugham, Saravanan wrote:
    =20
          > Are we assuming then that this is only limited to DTMF
          type-ahead and
          > is not speak ahead as well?
          >
          > Sarvi
          >
          >
          -----------------------------------------------------------
          -------------
          >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
          >     *Sent:* Tuesday, August 23, 2005 3:20 PM
          >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
          >     dburnett@vocalocity.net
          >     *Cc:* speechsc@ietf.org
          >     *Subject:* RE: [Speechsc] RE: Problem with DTMF=20
     type-ahead
          >
          >     It would seem like we're not trying to buffer audio
          (generally)
          >     but only the DIGIT values themselves.
          >
          >     It would seem better to just be a header such as:
          >
          >     DTMF-Buffer: <true|false>
          >
          >     instead.
          >
          >     Jeff
          >
          >
          -----------------------------------------------------------
          -------------
          >     *From:* speechsc-bounces@ietf.org
          >     [mailto:speechsc-bounces@ietf.org] *On Behalf=20
     Of *Shanmugham,
          >     Saravanan
          >     *Sent:* Tuesday, August 23, 2005 4:59 PM
          >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
          >     *Cc:* speechsc@ietf.org
          >     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
          >
          >
          >     I had typed this response out earlier but wanted to
          see if there
          >     was any support for this capability. So sending=20
     it out now.
          >
          >     The solution I would suggest is a new header
          >                Type-Ahead: <time-in-milli-seconds>
          >     header that can be sent on the SET-PARAMS,=20
     GET-PARAMS and
          >     RECOGNIZE methods.
          >
          >     default is 0 which is todays no buffering case. If
          set to some
          >     value, the resource is expected to buffer upto
          >     <time-in-milliseconds> of audio and use it during
          the recognize as
          >     a type-ahead buffer. I can see this being a boolean
          field as well,
          >     but the <time-in-milliseconds> approach is little
          more flexible.
          >
          >     Sarvi
          >
          >
          -----------------------------------------------------------
          -------------
          >         *From:* Reifenrath, Klaus
          [mailto:Klaus.Reifenrath@Scansoft.com]
          >         *Sent:* Tuesday, August 23, 2005 8:41 AM
          >         *To:* Shanmugham, Saravanan;=20
     'dburnett@vocalocity.net'
          >         *Cc:* 'speechsc@ietf.org'
          >         *Subject:* Problem with DTMF type-ahead
          >
          >         The MRCP and MRCPv2 specifications define=20
     how DTMF input
          >         should be handled: DTMF input is collected after
          the RECOGNIZE
          >         request is received until a term-char or timeout
          is reached.
          >         Then a result or no-match is returned based on
          any active DTMF
          >         grammars. The problem is there can be a
          significant amount of
          >         time between when one result is processed by the
          application
          >         and the next grammar is activated. During that
          time any DTMF
          >         input received will be lost. Such DTMF
          type-ahead is common
          >         for frequent users of IVR systems, and buffering
          that type
          >         ahead is required by VoiceXML 2.0 (see section
          4.1.8). Can
          >         this requirement of VoiceXML be implemented=20
     using MRCP
          >         (assuming that the media source CANNOT buffer
          DTMF)? If so,
          >         how is it done?
          >
          >         Thanks,
          >         Klaus
          >
          >----------------------------------------------------------
          -------------
          -
          >
          >_______________________________________________
          >Speechsc mailing list
          >Speechsc@ietf.org
          >https://www1.ietf.org/mailman/listinfo/speechsc
          >
          >
    =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
Date: Wed, 24 Aug 2005 14:39:29 -0400
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Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead
Thread-Index: AcWokY2nOWUMZ0/RTlajUyiwMSDepAADFCrAAA5tVzAAAK6aMA==
From: "Eric Burger" <eburger@brooktrout.com>
To: "Shanmugham, Saravanan" <sarvi@cisco.com>
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[not as chair]
I still see no use case for NOT having the buffer go back to the
beginning of time.  Since there is no need for a limit to the buffer
size, why complicate things?

Because someone will always come up with a bizarre corner case, KPML
does do a circular buffer thing, dropping real old tones, and indicating
that old tones were lost.  However, even on a memory constrained
telephone, that would be hours of digits being entered continuously
(i.e., still probably never could happen).

Thus, to the question of "Should the client set the buffer size?" the
answer is "no".  The size is effectively infinite.  "Is the size set by
the platform?"  I would say "no", because there is no reason to.
However, if you are ultra paranoid about a DoS attack that runs for a
really long time (which in itself is going to be the source of the
problem, not that your DTMF buffer overflows), I suppose one could say
"yes".

My personal opinion is still to just have an infinite buffer size, going
back to the last recognition / start of buffering / last flush.

-----Original Message-----
From: Shanmugham, Saravanan [mailto:sarvi@cisco.com]=20
Sent: Wednesday, August 24, 2005 2:22 PM
To: Eric Burger; Dave Burke
Cc: speechsc@ietf.org
Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead

Ok.
Lets limit this buffer to digits and not speech.=20
Lets also stick to clear buffer operation in RECOGNIZE method.

But should be the buffer size. Settable by the client? Platform specific
fixed? Settable client on a per RECOGNIZE basis.

Unlimited size may be ok for implementation, but is it practical and
what does it mean in actual usage.=20
If the recognizer channel was allocated at the beginning of the call,
and at some later point in the call, much later, we do a recognize. Does
this mean that any digits pressed since the beginning of the call when
the resource was allocated or the previous RECOGNIZE, are used as part
of the type-ahead?

Sarvi=20

     -----Original Message-----
     From: Eric Burger [mailto:eburger@brooktrout.com]=20
     Sent: Wednesday, August 24, 2005 4:31 AM
     To: Dave Burke; Shanmugham, Saravanan
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     [not as chair]
     We went through this whole discussion for KPML.  No one=20
     ever had a use case for "keep the last x seconds".  In=20
     fact, one either buffered digits or flushed them.
    =20
     If you have an 8,000 port media engine, with one hour call=20
     times, and continuous digit entry without the digits ever=20
     getting off the queue, and the digit entry is computer=20
     generated (i.e., as fast as possible), you need:
        (8,000 ports) * (3,600,000ms/port) / (60ms / digit) =3D 480MB
    =20
     I would offer that if you have an 8,000 port media=20
     processing device, less than half a gigabyte for digit=20
     buffering is nothing.
    =20
     In English, that means that you can realistically have=20
     infinite digit buffering.
    =20
     That makes the protocol MUCH easier, interoperation MUCH=20
     easier, and implementation MUCH easier.
    =20
     [really not as chair]
     As for the term, I didn't think anyone had an alternative=20
     interpretation of the term "flush".  However, I suppose=20
     "clear" is OK.
    =20
     -----Original Message-----
     From: Dave Burke [mailto:david.burke@voxpilot.com]
     Sent: Wednesday, August 24, 2005 5:52 AM
     To: Shanmugham, Saravanan; Eric Burger; speechsc@ietf.org
     Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     Sounds like a lot of dials to play with: What is the=20
     use-case warranting the extra complexity for overriding=20
     the SET-PARAMed buffer length on a per RECOGNIZE?
    =20
     The two capabilities that you want (IMO) are:
         1. Configuration: How long is the buffer?
         2. Action: Flush buffer
    =20
     For configuration, I'd rather see a SET-PARAMs but with a=20
     non-zero default - say 10 seconds (2 x default inter-digit=20
     time).  For flushing, I'd rather see a boolean header on=20
     RECOGNIZE for simplicity. That way, out-of-the-box, easy=20
     things are easy and harder things are still possible...
    =20
     I don't like the term flush. Is one flushing the DTMF into=20
     the recognizer or out into the ether(net)? Hence why I=20
     suggested Clear-DTMF-Buffer.
    =20
     Dave
    =20
     ----- Original Message -----
     From: "Shanmugham, Saravanan" <sarvi@cisco.com>
     To: "Eric Burger" <eburger@brooktrout.com>; <speechsc@ietf.org>
     Sent: Wednesday, August 24, 2005 7:43 AM
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
    =20
     The idea of being able to send it opn the RECOGNIZE is to=20
     reduce the
     time and not increase it.
    =20
     By default, the value is 0. When you do a SET-PARAMS and=20
     set it to say
     10000 it sets the buffering at the server to 10 seconds.
    =20
     This does not mean that every RECOGNIZE should use this=20
     entire 10s of
     buffer. On an individual RECOGNIZE method basis the client=20
     should be
     able to use 0 - 10s of that buffer. This would be the last=20
     X seconds of
     the buffer before the RECOGNIZE method is recieved the=20
     rest would be
     discarded.
    =20
     Achieves more flexibility than plain flushing true or false.
    =20
     Sarvi
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Eric Burger
          Sent: Tuesday, August 23, 2005 11:14 PM
          To: speechsc@ietf.org
          Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
          [not as chair]
    =20
          I can see something like "Type-Ahead" for SET/GET-PARAMS,
          which sets a global digit buffer.  I would suggest a more
          descriptive name, like DTMF-Buffer-Time.
    =20
          That said, I don't see it working for the RECOGNIZE
          method.  The speech resource needs to know *before* the
          RECOGNIZE request that it needs to buffer DTMF.  If it
          comes in with the RECOGNIZE request, it will be too late.
    =20
          For example:
    =20
          SET-PARAMS
          DTMF-Buffer-Time: 10000
          ...
          <sets global buffer to 10s; platform keeps 10 seconds of
          DTMF buffered>
    =20
    =20
          RECOGNIZE
          DTMF-Buffer-Time: 20000
          ...
          <sets buffer to 20s.  Oops: platform already dropped first
          10 seconds because it was only buffering 10 seconds>
    =20
    =20
    =20
          The sensible per-RECOGNIZE attribute is to flush the
          buffer.  While one might be tempted to say
          "DTMF-Buffer-Time: 0" means flush the buffer, I am sure
          someone will try to use it with a non-zero value.  I would
          offer we use a different attribute to indicate flushing
          the buffer before recognition, like "Flush-DTMF: true".
    =20
          -----Original Message-----
          From: speechsc-bounces@ietf.org
          [mailto:speechsc-bounces@ietf.org] On Behalf Of Corby Anderson
          Sent: Tuesday, August 23, 2005 7:34 PM
          To: speechsc@ietf.org
          Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
    =20
          I prefer the flexible "Type-Ahead: <time-in-milliseconds>"
          approach.
          This would be useful for implementing vxml listen states
          that happen after data tags.  If a data fetch takes longer
          than several seconds, you
    =20
          might not want to play dtmf that is older than some limit.
    =20
          Corby Anderson
          Tellme Networks, Inc.
    =20
          Shanmugham, Saravanan wrote:
    =20
          > Are we assuming then that this is only limited to DTMF
          type-ahead and
          > is not speak ahead as well?
          >
          > Sarvi
          >
          >
          -----------------------------------------------------------
          -------------
          >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
          >     *Sent:* Tuesday, August 23, 2005 3:20 PM
          >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
          >     dburnett@vocalocity.net
          >     *Cc:* speechsc@ietf.org
          >     *Subject:* RE: [Speechsc] RE: Problem with DTMF=20
     type-ahead
          >
          >     It would seem like we're not trying to buffer audio
          (generally)
          >     but only the DIGIT values themselves.
          >
          >     It would seem better to just be a header such as:
          >
          >     DTMF-Buffer: <true|false>
          >
          >     instead.
          >
          >     Jeff
          >
          >
          -----------------------------------------------------------
          -------------
          >     *From:* speechsc-bounces@ietf.org
          >     [mailto:speechsc-bounces@ietf.org] *On Behalf=20
     Of *Shanmugham,
          >     Saravanan
          >     *Sent:* Tuesday, August 23, 2005 4:59 PM
          >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
          >     *Cc:* speechsc@ietf.org
          >     *Subject:* [Speechsc] RE: Problem with DTMF type-ahead
          >
          >
          >     I had typed this response out earlier but wanted to
          see if there
          >     was any support for this capability. So sending=20
     it out now.
          >
          >     The solution I would suggest is a new header
          >                Type-Ahead: <time-in-milli-seconds>
          >     header that can be sent on the SET-PARAMS,=20
     GET-PARAMS and
          >     RECOGNIZE methods.
          >
          >     default is 0 which is todays no buffering case. If
          set to some
          >     value, the resource is expected to buffer upto
          >     <time-in-milliseconds> of audio and use it during
          the recognize as
          >     a type-ahead buffer. I can see this being a boolean
          field as well,
          >     but the <time-in-milliseconds> approach is little
          more flexible.
          >
          >     Sarvi
          >
          >
          -----------------------------------------------------------
          -------------
          >         *From:* Reifenrath, Klaus
          [mailto:Klaus.Reifenrath@Scansoft.com]
          >         *Sent:* Tuesday, August 23, 2005 8:41 AM
          >         *To:* Shanmugham, Saravanan;=20
     'dburnett@vocalocity.net'
          >         *Cc:* 'speechsc@ietf.org'
          >         *Subject:* Problem with DTMF type-ahead
          >
          >         The MRCP and MRCPv2 specifications define=20
     how DTMF input
          >         should be handled: DTMF input is collected after
          the RECOGNIZE
          >         request is received until a term-char or timeout
          is reached.
          >         Then a result or no-match is returned based on
          any active DTMF
          >         grammars. The problem is there can be a
          significant amount of
          >         time between when one result is processed by the
          application
          >         and the next grammar is activated. During that
          time any DTMF
          >         input received will be lost. Such DTMF
          type-ahead is common
          >         for frequent users of IVR systems, and buffering
          that type
          >         ahead is required by VoiceXML 2.0 (see section
          4.1.8). Can
          >         this requirement of VoiceXML be implemented=20
     using MRCP
          >         (assuming that the media source CANNOT buffer
          DTMF)? If so,
          >         how is it done?
          >
          >         Thanks,
          >         Klaus
          >
          >----------------------------------------------------------
          -------------
          -
          >
          >_______________________________________________
          >Speechsc mailing list
          >Speechsc@ietf.org
          >https://www1.ietf.org/mailman/listinfo/speechsc
          >
          >
    =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
    =20
    =20
     _______________________________________________
     Speechsc mailing list
     Speechsc@ietf.org
     https://www1.ietf.org/mailman/listinfo/speechsc
    =20

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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
Date: Wed, 24 Aug 2005 12:56:46 -0700
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Thread-Topic: [Speechsc] RE: Problem with DTMF type-ahead
Thread-Index: AcWokY2nOWUMZ0/RTlajUyiwMSDepAADFCrAAA5tVzAAAK6aMAACiJ4A
From: "Shanmugham, Saravanan" <sarvi@cisco.com>
To: "Eric Burger" <eburger@brooktrout.com>
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I am less concerned about DoS attacks.=20
We are implementing a recognition resource that could potentially be
attached a=20
Speaker for the entire period of the call, without actually doing any
actual=20
Recognition. Soemtime way into the call the application might play a
prompt and start a Recognize. The question is will any digits pressed
during the call be applied because the buffer was too long. Note that we
are actually saying the resource would be buffering digits even when
there is no Recognize happenning.

Anyway, If there is much interest in the group for supporting variable
buffer size, Iwill not push it.

Sarvi

     -----Original Message-----
     From: Eric Burger [mailto:eburger@brooktrout.com]=20
     Sent: Wednesday, August 24, 2005 11:39 AM
     To: Shanmugham, Saravanan
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     [not as chair]
     I still see no use case for NOT having the buffer go back=20
     to the beginning of time.  Since there is no need for a=20
     limit to the buffer size, why complicate things?
    =20
     Because someone will always come up with a bizarre corner=20
     case, KPML does do a circular buffer thing, dropping real=20
     old tones, and indicating that old tones were lost. =20
     However, even on a memory constrained telephone, that=20
     would be hours of digits being entered continuously (i.e.,=20
     still probably never could happen).
    =20
     Thus, to the question of "Should the client set the buffer=20
     size?" the answer is "no".  The size is effectively=20
     infinite.  "Is the size set by the platform?"  I would say=20
     "no", because there is no reason to.
     However, if you are ultra paranoid about a DoS attack that=20
     runs for a really long time (which in itself is going to=20
     be the source of the problem, not that your DTMF buffer=20
     overflows), I suppose one could say "yes".
    =20
     My personal opinion is still to just have an infinite=20
     buffer size, going back to the last recognition / start of=20
     buffering / last flush.
    =20
     -----Original Message-----
     From: Shanmugham, Saravanan [mailto:sarvi@cisco.com]
     Sent: Wednesday, August 24, 2005 2:22 PM
     To: Eric Burger; Dave Burke
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     Ok.
     Lets limit this buffer to digits and not speech.=20
     Lets also stick to clear buffer operation in RECOGNIZE method.
    =20
     But should be the buffer size. Settable by the client?=20
     Platform specific fixed? Settable client on a per RECOGNIZE basis.
    =20
     Unlimited size may be ok for implementation, but is it=20
     practical and what does it mean in actual usage.=20
     If the recognizer channel was allocated at the beginning=20
     of the call, and at some later point in the call, much=20
     later, we do a recognize. Does this mean that any digits=20
     pressed since the beginning of the call when the resource=20
     was allocated or the previous RECOGNIZE, are used as part=20
     of the type-ahead?
    =20
     Sarvi=20
    =20
          -----Original Message-----
          From: Eric Burger [mailto:eburger@brooktrout.com]=20
          Sent: Wednesday, August 24, 2005 4:31 AM
          To: Dave Burke; Shanmugham, Saravanan
          Cc: speechsc@ietf.org
          Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
         =20
          [not as chair]
          We went through this whole discussion for KPML.  No one=20
          ever had a use case for "keep the last x seconds".  In=20
          fact, one either buffered digits or flushed them.
         =20
          If you have an 8,000 port media engine, with one hour call=20
          times, and continuous digit entry without the digits ever=20
          getting off the queue, and the digit entry is computer=20
          generated (i.e., as fast as possible), you need:
             (8,000 ports) * (3,600,000ms/port) / (60ms / digit) =3D =
480MB
         =20
          I would offer that if you have an 8,000 port media=20
          processing device, less than half a gigabyte for digit=20
          buffering is nothing.
         =20
          In English, that means that you can realistically have=20
          infinite digit buffering.
         =20
          That makes the protocol MUCH easier, interoperation MUCH=20
          easier, and implementation MUCH easier.
         =20
          [really not as chair]
          As for the term, I didn't think anyone had an alternative=20
          interpretation of the term "flush".  However, I suppose=20
          "clear" is OK.
         =20
          -----Original Message-----
          From: Dave Burke [mailto:david.burke@voxpilot.com]
          Sent: Wednesday, August 24, 2005 5:52 AM
          To: Shanmugham, Saravanan; Eric Burger; speechsc@ietf.org
          Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
         =20
          Sounds like a lot of dials to play with: What is the=20
          use-case warranting the extra complexity for overriding=20
          the SET-PARAMed buffer length on a per RECOGNIZE?
         =20
          The two capabilities that you want (IMO) are:
              1. Configuration: How long is the buffer?
              2. Action: Flush buffer
         =20
          For configuration, I'd rather see a SET-PARAMs but with a=20
          non-zero default - say 10 seconds (2 x default inter-digit=20
          time).  For flushing, I'd rather see a boolean header on=20
          RECOGNIZE for simplicity. That way, out-of-the-box, easy=20
          things are easy and harder things are still possible...
         =20
          I don't like the term flush. Is one flushing the DTMF into=20
          the recognizer or out into the ether(net)? Hence why I=20
          suggested Clear-DTMF-Buffer.
         =20
          Dave
         =20
          ----- Original Message -----
          From: "Shanmugham, Saravanan" <sarvi@cisco.com>
          To: "Eric Burger" <eburger@brooktrout.com>;=20
     <speechsc@ietf.org>
          Sent: Wednesday, August 24, 2005 7:43 AM
          Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
         =20
         =20
          The idea of being able to send it opn the RECOGNIZE is to=20
          reduce the
          time and not increase it.
         =20
          By default, the value is 0. When you do a SET-PARAMS and=20
          set it to say
          10000 it sets the buffering at the server to 10 seconds.
         =20
          This does not mean that every RECOGNIZE should use this=20
          entire 10s of
          buffer. On an individual RECOGNIZE method basis the client=20
          should be
          able to use 0 - 10s of that buffer. This would be the last=20
          X seconds of
          the buffer before the RECOGNIZE method is recieved the=20
          rest would be
          discarded.
         =20
          Achieves more flexibility than plain flushing true or false.
         =20
          Sarvi
         =20
               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Eric Burger
               Sent: Tuesday, August 23, 2005 11:14 PM
               To: speechsc@ietf.org
               Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
         =20
               [not as chair]
         =20
               I can see something like "Type-Ahead" for SET/GET-PARAMS,
               which sets a global digit buffer.  I would suggest a more
               descriptive name, like DTMF-Buffer-Time.
         =20
               That said, I don't see it working for the RECOGNIZE
               method.  The speech resource needs to know *before* the
               RECOGNIZE request that it needs to buffer DTMF.  If it
               comes in with the RECOGNIZE request, it will be too late.
         =20
               For example:
         =20
               SET-PARAMS
               DTMF-Buffer-Time: 10000
               ...
               <sets global buffer to 10s; platform keeps 10 seconds of
               DTMF buffered>
         =20
         =20
               RECOGNIZE
               DTMF-Buffer-Time: 20000
               ...
               <sets buffer to 20s.  Oops: platform already=20
     dropped first
               10 seconds because it was only buffering 10 seconds>
         =20
         =20
         =20
               The sensible per-RECOGNIZE attribute is to flush the
               buffer.  While one might be tempted to say
               "DTMF-Buffer-Time: 0" means flush the buffer, I am sure
               someone will try to use it with a non-zero=20
     value.  I would
               offer we use a different attribute to indicate flushing
               the buffer before recognition, like "Flush-DTMF: true".
         =20
               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Corby Anderson
               Sent: Tuesday, August 23, 2005 7:34 PM
               To: speechsc@ietf.org
               Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
         =20
               I prefer the flexible "Type-Ahead:=20
     <time-in-milliseconds>"
               approach.
               This would be useful for implementing vxml listen states
               that happen after data tags.  If a data fetch=20
     takes longer
               than several seconds, you
         =20
               might not want to play dtmf that is older than=20
     some limit.
         =20
               Corby Anderson
               Tellme Networks, Inc.
         =20
               Shanmugham, Saravanan wrote:
         =20
               > Are we assuming then that this is only limited to DTMF
               type-ahead and
               > is not speak ahead as well?
               >
               > Sarvi
               >
               >
              =20
     -----------------------------------------------------------
               -------------
               >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
               >     *Sent:* Tuesday, August 23, 2005 3:20 PM
               >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
               >     dburnett@vocalocity.net
               >     *Cc:* speechsc@ietf.org
               >     *Subject:* RE: [Speechsc] RE: Problem with DTMF=20
          type-ahead
               >
               >     It would seem like we're not trying to buffer audio
               (generally)
               >     but only the DIGIT values themselves.
               >
               >     It would seem better to just be a header such as:
               >
               >     DTMF-Buffer: <true|false>
               >
               >     instead.
               >
               >     Jeff
               >
               >
              =20
     -----------------------------------------------------------
               -------------
               >     *From:* speechsc-bounces@ietf.org
               >     [mailto:speechsc-bounces@ietf.org] *On Behalf=20
          Of *Shanmugham,
               >     Saravanan
               >     *Sent:* Tuesday, August 23, 2005 4:59 PM
               >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
               >     *Cc:* speechsc@ietf.org
               >     *Subject:* [Speechsc] RE: Problem with=20
     DTMF type-ahead
               >
               >
               >     I had typed this response out earlier but wanted to
               see if there
               >     was any support for this capability. So sending=20
          it out now.
               >
               >     The solution I would suggest is a new header
               >                Type-Ahead: <time-in-milli-seconds>
               >     header that can be sent on the SET-PARAMS,=20
          GET-PARAMS and
               >     RECOGNIZE methods.
               >
               >     default is 0 which is todays no buffering case. If
               set to some
               >     value, the resource is expected to buffer upto
               >     <time-in-milliseconds> of audio and use it during
               the recognize as
               >     a type-ahead buffer. I can see this being a boolean
               field as well,
               >     but the <time-in-milliseconds> approach is little
               more flexible.
               >
               >     Sarvi
               >
               >
              =20
     -----------------------------------------------------------
               -------------
               >         *From:* Reifenrath, Klaus
               [mailto:Klaus.Reifenrath@Scansoft.com]
               >         *Sent:* Tuesday, August 23, 2005 8:41 AM
               >         *To:* Shanmugham, Saravanan;=20
          'dburnett@vocalocity.net'
               >         *Cc:* 'speechsc@ietf.org'
               >         *Subject:* Problem with DTMF type-ahead
               >
               >         The MRCP and MRCPv2 specifications define=20
          how DTMF input
               >         should be handled: DTMF input is=20
     collected after
               the RECOGNIZE
               >         request is received until a term-char=20
     or timeout
               is reached.
               >         Then a result or no-match is returned based on
               any active DTMF
               >         grammars. The problem is there can be a
               significant amount of
               >         time between when one result is=20
     processed by the
               application
               >         and the next grammar is activated. During that
               time any DTMF
               >         input received will be lost. Such DTMF
               type-ahead is common
               >         for frequent users of IVR systems, and=20
     buffering
               that type
               >         ahead is required by VoiceXML 2.0 (see section
               4.1.8). Can
               >         this requirement of VoiceXML be implemented=20
          using MRCP
               >         (assuming that the media source CANNOT buffer
               DTMF)? If so,
               >         how is it done?
               >
               >         Thanks,
               >         Klaus
               >
              =20
     >----------------------------------------------------------
               -------------
               -
               >
               >_______________________________________________
               >Speechsc mailing list
               >Speechsc@ietf.org
               >https://www1.ietf.org/mailman/listinfo/speechsc
               >
               >
         =20
               _______________________________________________
               Speechsc mailing list
               Speechsc@ietf.org
               https://www1.ietf.org/mailman/listinfo/speechsc
         =20
               _______________________________________________
               Speechsc mailing list
               Speechsc@ietf.org
               https://www1.ietf.org/mailman/listinfo/speechsc
         =20
         =20
          _______________________________________________
          Speechsc mailing list
          Speechsc@ietf.org
          https://www1.ietf.org/mailman/listinfo/speechsc
         =20
    =20

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Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
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Having variable or infinite buffering would not change the client
behavior in the example given below.

At some point, the client says, "I am interested in digits starting
NOW."  The "NOW" point is where the client flushes the buffer.  Anything
else would not be deterministic.

-----Original Message-----
From: Shanmugham, Saravanan [mailto:sarvi@cisco.com]=20
Sent: Wednesday, August 24, 2005 3:57 PM
To: Eric Burger
Cc: speechsc@ietf.org
Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead

I am less concerned about DoS attacks.=20
We are implementing a recognition resource that could potentially be
attached a=20
Speaker for the entire period of the call, without actually doing any
actual=20
Recognition. Soemtime way into the call the application might play a
prompt and start a Recognize. The question is will any digits pressed
during the call be applied because the buffer was too long. Note that we
are actually saying the resource would be buffering digits even when
there is no Recognize happenning.

Anyway, If there is much interest in the group for supporting variable
buffer size, Iwill not push it.

Sarvi

     -----Original Message-----
     From: Eric Burger [mailto:eburger@brooktrout.com]=20
     Sent: Wednesday, August 24, 2005 11:39 AM
     To: Shanmugham, Saravanan
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     [not as chair]
     I still see no use case for NOT having the buffer go back=20
     to the beginning of time.  Since there is no need for a=20
     limit to the buffer size, why complicate things?
    =20
     Because someone will always come up with a bizarre corner=20
     case, KPML does do a circular buffer thing, dropping real=20
     old tones, and indicating that old tones were lost. =20
     However, even on a memory constrained telephone, that=20
     would be hours of digits being entered continuously (i.e.,=20
     still probably never could happen).
    =20
     Thus, to the question of "Should the client set the buffer=20
     size?" the answer is "no".  The size is effectively=20
     infinite.  "Is the size set by the platform?"  I would say=20
     "no", because there is no reason to.
     However, if you are ultra paranoid about a DoS attack that=20
     runs for a really long time (which in itself is going to=20
     be the source of the problem, not that your DTMF buffer=20
     overflows), I suppose one could say "yes".
    =20
     My personal opinion is still to just have an infinite=20
     buffer size, going back to the last recognition / start of=20
     buffering / last flush.
    =20
     -----Original Message-----
     From: Shanmugham, Saravanan [mailto:sarvi@cisco.com]
     Sent: Wednesday, August 24, 2005 2:22 PM
     To: Eric Burger; Dave Burke
     Cc: speechsc@ietf.org
     Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
    =20
     Ok.
     Lets limit this buffer to digits and not speech.=20
     Lets also stick to clear buffer operation in RECOGNIZE method.
    =20
     But should be the buffer size. Settable by the client?=20
     Platform specific fixed? Settable client on a per RECOGNIZE basis.
    =20
     Unlimited size may be ok for implementation, but is it=20
     practical and what does it mean in actual usage.=20
     If the recognizer channel was allocated at the beginning=20
     of the call, and at some later point in the call, much=20
     later, we do a recognize. Does this mean that any digits=20
     pressed since the beginning of the call when the resource=20
     was allocated or the previous RECOGNIZE, are used as part=20
     of the type-ahead?
    =20
     Sarvi=20
    =20
          -----Original Message-----
          From: Eric Burger [mailto:eburger@brooktrout.com]=20
          Sent: Wednesday, August 24, 2005 4:31 AM
          To: Dave Burke; Shanmugham, Saravanan
          Cc: speechsc@ietf.org
          Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
         =20
          [not as chair]
          We went through this whole discussion for KPML.  No one=20
          ever had a use case for "keep the last x seconds".  In=20
          fact, one either buffered digits or flushed them.
         =20
          If you have an 8,000 port media engine, with one hour call=20
          times, and continuous digit entry without the digits ever=20
          getting off the queue, and the digit entry is computer=20
          generated (i.e., as fast as possible), you need:
             (8,000 ports) * (3,600,000ms/port) / (60ms / digit) =3D =
480MB
         =20
          I would offer that if you have an 8,000 port media=20
          processing device, less than half a gigabyte for digit=20
          buffering is nothing.
         =20
          In English, that means that you can realistically have=20
          infinite digit buffering.
         =20
          That makes the protocol MUCH easier, interoperation MUCH=20
          easier, and implementation MUCH easier.
         =20
          [really not as chair]
          As for the term, I didn't think anyone had an alternative=20
          interpretation of the term "flush".  However, I suppose=20
          "clear" is OK.
         =20
          -----Original Message-----
          From: Dave Burke [mailto:david.burke@voxpilot.com]
          Sent: Wednesday, August 24, 2005 5:52 AM
          To: Shanmugham, Saravanan; Eric Burger; speechsc@ietf.org
          Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
         =20
          Sounds like a lot of dials to play with: What is the=20
          use-case warranting the extra complexity for overriding=20
          the SET-PARAMed buffer length on a per RECOGNIZE?
         =20
          The two capabilities that you want (IMO) are:
              1. Configuration: How long is the buffer?
              2. Action: Flush buffer
         =20
          For configuration, I'd rather see a SET-PARAMs but with a=20
          non-zero default - say 10 seconds (2 x default inter-digit=20
          time).  For flushing, I'd rather see a boolean header on=20
          RECOGNIZE for simplicity. That way, out-of-the-box, easy=20
          things are easy and harder things are still possible...
         =20
          I don't like the term flush. Is one flushing the DTMF into=20
          the recognizer or out into the ether(net)? Hence why I=20
          suggested Clear-DTMF-Buffer.
         =20
          Dave
         =20
          ----- Original Message -----
          From: "Shanmugham, Saravanan" <sarvi@cisco.com>
          To: "Eric Burger" <eburger@brooktrout.com>;=20
     <speechsc@ietf.org>
          Sent: Wednesday, August 24, 2005 7:43 AM
          Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
         =20
         =20
          The idea of being able to send it opn the RECOGNIZE is to=20
          reduce the
          time and not increase it.
         =20
          By default, the value is 0. When you do a SET-PARAMS and=20
          set it to say
          10000 it sets the buffering at the server to 10 seconds.
         =20
          This does not mean that every RECOGNIZE should use this=20
          entire 10s of
          buffer. On an individual RECOGNIZE method basis the client=20
          should be
          able to use 0 - 10s of that buffer. This would be the last=20
          X seconds of
          the buffer before the RECOGNIZE method is recieved the=20
          rest would be
          discarded.
         =20
          Achieves more flexibility than plain flushing true or false.
         =20
          Sarvi
         =20
               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Eric Burger
               Sent: Tuesday, August 23, 2005 11:14 PM
               To: speechsc@ietf.org
               Subject: RE: [Speechsc] RE: Problem with DTMF type-ahead
         =20
               [not as chair]
         =20
               I can see something like "Type-Ahead" for SET/GET-PARAMS,
               which sets a global digit buffer.  I would suggest a more
               descriptive name, like DTMF-Buffer-Time.
         =20
               That said, I don't see it working for the RECOGNIZE
               method.  The speech resource needs to know *before* the
               RECOGNIZE request that it needs to buffer DTMF.  If it
               comes in with the RECOGNIZE request, it will be too late.
         =20
               For example:
         =20
               SET-PARAMS
               DTMF-Buffer-Time: 10000
               ...
               <sets global buffer to 10s; platform keeps 10 seconds of
               DTMF buffered>
         =20
         =20
               RECOGNIZE
               DTMF-Buffer-Time: 20000
               ...
               <sets buffer to 20s.  Oops: platform already=20
     dropped first
               10 seconds because it was only buffering 10 seconds>
         =20
         =20
         =20
               The sensible per-RECOGNIZE attribute is to flush the
               buffer.  While one might be tempted to say
               "DTMF-Buffer-Time: 0" means flush the buffer, I am sure
               someone will try to use it with a non-zero=20
     value.  I would
               offer we use a different attribute to indicate flushing
               the buffer before recognition, like "Flush-DTMF: true".
         =20
               -----Original Message-----
               From: speechsc-bounces@ietf.org
               [mailto:speechsc-bounces@ietf.org] On Behalf Of=20
     Corby Anderson
               Sent: Tuesday, August 23, 2005 7:34 PM
               To: speechsc@ietf.org
               Subject: Re: [Speechsc] RE: Problem with DTMF type-ahead
         =20
               I prefer the flexible "Type-Ahead:=20
     <time-in-milliseconds>"
               approach.
               This would be useful for implementing vxml listen states
               that happen after data tags.  If a data fetch=20
     takes longer
               than several seconds, you
         =20
               might not want to play dtmf that is older than=20
     some limit.
         =20
               Corby Anderson
               Tellme Networks, Inc.
         =20
               Shanmugham, Saravanan wrote:
         =20
               > Are we assuming then that this is only limited to DTMF
               type-ahead and
               > is not speak ahead as well?
               >
               > Sarvi
               >
               >
              =20
     -----------------------------------------------------------
               -------------
               >     *From:* Jeff Haynie [mailto:jhaynie@vocalocity.net]
               >     *Sent:* Tuesday, August 23, 2005 3:20 PM
               >     *To:* Shanmugham, Saravanan; Reifenrath, Klaus;
               >     dburnett@vocalocity.net
               >     *Cc:* speechsc@ietf.org
               >     *Subject:* RE: [Speechsc] RE: Problem with DTMF=20
          type-ahead
               >
               >     It would seem like we're not trying to buffer audio
               (generally)
               >     but only the DIGIT values themselves.
               >
               >     It would seem better to just be a header such as:
               >
               >     DTMF-Buffer: <true|false>
               >
               >     instead.
               >
               >     Jeff
               >
               >
              =20
     -----------------------------------------------------------
               -------------
               >     *From:* speechsc-bounces@ietf.org
               >     [mailto:speechsc-bounces@ietf.org] *On Behalf=20
          Of *Shanmugham,
               >     Saravanan
               >     *Sent:* Tuesday, August 23, 2005 4:59 PM
               >     *To:* Reifenrath, Klaus; dburnett@vocalocity.net
               >     *Cc:* speechsc@ietf.org
               >     *Subject:* [Speechsc] RE: Problem with=20
     DTMF type-ahead
               >
               >
               >     I had typed this response out earlier but wanted to
               see if there
               >     was any support for this capability. So sending=20
          it out now.
               >
               >     The solution I would suggest is a new header
               >                Type-Ahead: <time-in-milli-seconds>
               >     header that can be sent on the SET-PARAMS,=20
          GET-PARAMS and
               >     RECOGNIZE methods.
               >
               >     default is 0 which is todays no buffering case. If
               set to some
               >     value, the resource is expected to buffer upto
               >     <time-in-milliseconds> of audio and use it during
               the recognize as
               >     a type-ahead buffer. I can see this being a boolean
               field as well,
               >     but the <time-in-milliseconds> approach is little
               more flexible.
               >
               >     Sarvi
               >
               >
              =20
     -----------------------------------------------------------
               -------------
               >         *From:* Reifenrath, Klaus
               [mailto:Klaus.Reifenrath@Scansoft.com]
               >         *Sent:* Tuesday, August 23, 2005 8:41 AM
               >         *To:* Shanmugham, Saravanan;=20
          'dburnett@vocalocity.net'
               >         *Cc:* 'speechsc@ietf.org'
               >         *Subject:* Problem with DTMF type-ahead
               >
               >         The MRCP and MRCPv2 specifications define=20
          how DTMF input
               >         should be handled: DTMF input is=20
     collected after
               the RECOGNIZE
               >         request is received until a term-char=20
     or timeout
               is reached.
               >         Then a result or no-match is returned based on
               any active DTMF
               >         grammars. The problem is there can be a
               significant amount of
               >         time between when one result is=20
     processed by the
               application
               >         and the next grammar is activated. During that
               time any DTMF
               >         input received will be lost. Such DTMF
               type-ahead is common
               >         for frequent users of IVR systems, and=20
     buffering
               that type
               >         ahead is required by VoiceXML 2.0 (see section
               4.1.8). Can
               >         this requirement of VoiceXML be implemented=20
          using MRCP
               >         (assuming that the media source CANNOT buffer
               DTMF)? If so,
               >         how is it done?
               >
               >         Thanks,
               >         Klaus
               >
              =20
     >----------------------------------------------------------
               -------------
               -
               >
               >_______________________________________________
               >Speechsc mailing list
               >Speechsc@ietf.org
               >https://www1.ietf.org/mailman/listinfo/speechsc
               >
               >
         =20
               _______________________________________________
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         =20
               _______________________________________________
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         =20
         =20
          _______________________________________________
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         =20
    =20

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From speechsc-bounces@ietf.org Mon Aug 29 10:34:31 2005
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Date: Mon, 29 Aug 2005 16:34:04 +0200
From: =?ISO-8859-1?Q?Jes=FAs_Bernat_Vercher?= <bernat@tid.es>
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Subject: [Speechsc] Question about Contiguous Request-ID's
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We already had a discussion on this issue
(http://www1.ietf.org/mail-archive/web/speechsc/current/msg01260.html)

It ended saying to change the draft to define the request-id as 
monotonically increasing and to define some errors:

"So, the above is the long way of saying I think we should do (2), and 
reject duplicate and out-of-order requests with a well-defined error. "

Is there any specific error in the currect draft for duplicated or 
out-of-order request-id's?

Thanks:

Jesus.


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