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From: "Brian Eberman" <bse@speechworks.com>
To: "Eric Burger" <eburger@snowshore.com>,
        "IETF SPEECHSC \(E-mail\)" <speechsc@ietf.org>
Subject: RE: [Speechsc] Administrative Fiat
Date: Mon, 5 May 2003 15:12:46 -0400
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Eric,

Seems to be a bit of a consensus here. What are the next steps to be
completed?

-Brian E.

Director Product Management and Services
SpeechWorks International
695 Atlantic Avenue
Boston, MA 02111
617 428 4444


-----Original Message-----
From: speechsc-admin@ietf.org [mailto:speechsc-admin@ietf.org]On Behalf
Of Rajiv Dharmadhikari
Sent: Friday, April 25, 2003 12:51 PM
To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Brian,

Can you resend the final draft of the protocl evaluation analysis to the
initial team? What is left to be completed? Let's get it out of the way..=
 I
am willing do complete my part since I don't know BEEP, COBOL and CORBA :=
-)

Thanks,

-Rajiv

-----Original Message-----
From: Brian Wyld [mailto:brian.wyld@eloquant.com]
Sent: Friday, April 25, 2003 1:46 AM
To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Sounds good to me...... although I believe there is still a significant b=
ase
of COBOL applications that may be critical to address.......

Or, I contra-propose:
1. the protocol will be RTSP extended to include the current MRCP message=
s
'natively', with addition of specific  messages for SR/SI, and to define =
the
'call session' mapping to the MRCP requests.

2. there will be a web services binding....

(sorry to be serious here)

Brian

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

> -----Message d'origine-----
> De : speechsc-admin@ietf.org
> [mailto:speechsc-admin@ietf.org]De la part
> de Eric Burger
> Envoy=E9 : Thursday, April 24, 2003 17:03
> =C0 : IETF SPEECHSC (E-mail)
> Objet : [Speechsc] Administrative Fiat
>
>
> Date: 1 April 2003
> [everything associated with speechsc seems to go out late]
>
> Since there has been no progress on the protocol analysis
> document, I have made the following decisions:
>
> 1. We need a new protocol.
>
> 2. Since no one seems to care, the protocol will be based on
> BEEP and CORBA.
>
> 3. To ensure that people use the protocol, we will publish an
> ADA binding.
>
> Unless I hear otherwise, this is the direction we will go.
>
> --
> - Eric
>
>
> P.S.  If I don't hear otherwise, we can consider the work of
> the work group complete.
>
> _______________________________________________
> 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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FN:Brian Eberman
ORG:Speechworks International;Product Development
TITLE:Director of Engineering, Carrier Technologies and Platforms
TEL;WORK;VOICE:(617) 428-4444
TEL;WORK;FAX:617 428 1122
ADR;WORK;ENCODING=3DQUOTED-PRINTABLE:;617 428 4444;695 Atlantic =
Ave=3D0D=3D0A;Boston;MA;02111
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Atlantic Ave=3D0D=3D0A=3D0D=3D0ABoston, MA 02111
URL;WORK:http://www.speechworks.com
EMAIL;PREF;INTERNET:bse@speechworks.com
EMAIL;INTERNET:beberman@bellsouthips.com
REV:20020505T175243Z
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_______________________________________________
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Thread-Topic: [Speechsc] Administrative Fiat
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From: "Eric Burger" <eburger@snowshore.com>
To: "Brian Eberman" <bse@speechworks.com>,
        "IETF SPEECHSC (E-mail)" <speechsc@ietf.org>
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Subject: [Speechsc] speechsc as a first-level protocol
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Is there consensus on raising speechsc to being a first-level protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?

Does anyone strongly object?

Would anyone be really upset if it isn't?

Is there value to it being a first-level protocol inside an existing protocol?  That is, use the session establishment and proxy infrastructure of SIP or RTSP and have speechsc-specific primitives as protocol extensions.

> -----Original Message-----
> From: Brian Eberman [mailto:bse@speechworks.com]
> Sent: Mon, May 05, 2003 3:13 PM
> To: Eric Burger; IETF SPEECHSC (E-mail)
> Subject: RE: [Speechsc] Administrative Fiat
> 
> 
> Eric,
> 
> Seems to be a bit of a consensus here. What are the next steps to be
> completed?
> 
> -Brian E.
> 
> Director Product Management and Services
> SpeechWorks International
> 695 Atlantic Avenue
> Boston, MA 02111
> 617 428 4444
> 
> 
> -----Original Message-----
> From: speechsc-admin@ietf.org 
> [mailto:speechsc-admin@ietf.org]On Behalf
> Of Rajiv Dharmadhikari
> Sent: Friday, April 25, 2003 12:51 PM
> To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
> Subject: RE: [Speechsc] Administrative Fiat
> 
> 
> Brian,
> 
> Can you resend the final draft of the protocl evaluation 
> analysis to the
> initial team? What is left to be completed? Let's get it out 
> of the way.. I
> am willing do complete my part since I don't know BEEP, COBOL 
> and CORBA :-)
> 
> Thanks,
> 
> -Rajiv
> 
> -----Original Message-----
> From: Brian Wyld [mailto:brian.wyld@eloquant.com]
> Sent: Friday, April 25, 2003 1:46 AM
> To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
> Subject: RE: [Speechsc] Administrative Fiat
> 
> 
> Eric,
> 
> Sounds good to me...... although I believe there is still a 
> significant base
> of COBOL applications that may be critical to address.......
> 
> Or, I contra-propose:
> 1. the protocol will be RTSP extended to include the current 
> MRCP messages
> 'natively', with addition of specific  messages for SR/SI, 
> and to define the
> 'call session' mapping to the MRCP requests.
> 
> 2. there will be a web services binding....
> 
> (sorry to be serious here)
> 
> Brian
> 
> [Brian Wyld] [brian.wyld@eloquant.com]
> [Directeur General R&D]
> [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
> [advanced solutions for telecoms and IT services]
> 
> > -----Message d'origine-----
> > De : speechsc-admin@ietf.org
> > [mailto:speechsc-admin@ietf.org]De la part
> > de Eric Burger
> > Envoyé : Thursday, April 24, 2003 17:03
> > À : IETF SPEECHSC (E-mail)
> > Objet : [Speechsc] Administrative Fiat
> >
> >
> > Date: 1 April 2003
> > [everything associated with speechsc seems to go out late]
> >
> > Since there has been no progress on the protocol analysis
> > document, I have made the following decisions:
> >
> > 1. We need a new protocol.
> >
> > 2. Since no one seems to care, the protocol will be based on
> > BEEP and CORBA.
> >
> > 3. To ensure that people use the protocol, we will publish an
> > ADA binding.
> >
> > Unless I hear otherwise, this is the direction we will go.
> >
> > --
> > - Eric
> >
> >
> > P.S.  If I don't hear otherwise, we can consider the work of
> > the work group complete.
> >
> > _______________________________________________
> > 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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Cc: "Eric Burger" <eburger@snowshore.com>,
        "Brian Eberman" <bse@speechworks.com>
Subject: RE: [Speechsc] speechsc as a first-level protocol
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> Is there value to it being a first-level protocol inside an existing
protocol?

Yes technically.

However, pragmatically, the difficulty for users/consumers is
getting a usable implementation from the open/free source
arena that is free from vendor drag.  This requires forethought
about how current open source efforts might fit.  Ok so I'm
putting the cart before the horse but its worth some thought
given the complexity of existing VoIP implementations which are
available. And where complexity could be a real barrier to adoption.

Thanks
Karl

-----Original Message-----
From: speechsc-admin@ietf.org [mailto:speechsc-admin@ietf.org]On Behalf
Of Eric Burger
Sent: Thursday, May 08, 2003 11:21 AM
To: Brian Eberman; IETF SPEECHSC (E-mail)
Subject: [Speechsc] speechsc as a first-level protocol


Is there consensus on raising speechsc to being a first-level protocol?
That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?

Does anyone strongly object?

Would anyone be really upset if it isn't?

Is there value to it being a first-level protocol inside an existing
protocol?  That is, use the session establishment and proxy infrastructure
of SIP or RTSP and have speechsc-specific primitives as protocol extensions.

> -----Original Message-----
> From: Brian Eberman [mailto:bse@speechworks.com]
> Sent: Mon, May 05, 2003 3:13 PM
> To: Eric Burger; IETF SPEECHSC (E-mail)
> Subject: RE: [Speechsc] Administrative Fiat
>
>
> Eric,
>
> Seems to be a bit of a consensus here. What are the next steps to be
> completed?
>
> -Brian E.
>
> Director Product Management and Services
> SpeechWorks International
> 695 Atlantic Avenue
> Boston, MA 02111
> 617 428 4444
>
>
> -----Original Message-----
> From: speechsc-admin@ietf.org
> [mailto:speechsc-admin@ietf.org]On Behalf
> Of Rajiv Dharmadhikari
> Sent: Friday, April 25, 2003 12:51 PM
> To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
> Subject: RE: [Speechsc] Administrative Fiat
>
>
> Brian,
>
> Can you resend the final draft of the protocl evaluation
> analysis to the
> initial team? What is left to be completed? Let's get it out
> of the way.. I
> am willing do complete my part since I don't know BEEP, COBOL
> and CORBA :-)
>
> Thanks,
>
> -Rajiv
>
> -----Original Message-----
> From: Brian Wyld [mailto:brian.wyld@eloquant.com]
> Sent: Friday, April 25, 2003 1:46 AM
> To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
> Subject: RE: [Speechsc] Administrative Fiat
>
>
> Eric,
>
> Sounds good to me...... although I believe there is still a
> significant base
> of COBOL applications that may be critical to address.......
>
> Or, I contra-propose:
> 1. the protocol will be RTSP extended to include the current
> MRCP messages
> 'natively', with addition of specific  messages for SR/SI,
> and to define the
> 'call session' mapping to the MRCP requests.
>
> 2. there will be a web services binding....
>
> (sorry to be serious here)
>
> Brian
>
> [Brian Wyld] [brian.wyld@eloquant.com]
> [Directeur General R&D]
> [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
> [advanced solutions for telecoms and IT services]
>
> > -----Message d'origine-----
> > De : speechsc-admin@ietf.org
> > [mailto:speechsc-admin@ietf.org]De la part
> > de Eric Burger
> > Envoyé : Thursday, April 24, 2003 17:03
> > À : IETF SPEECHSC (E-mail)
> > Objet : [Speechsc] Administrative Fiat
> >
> >
> > Date: 1 April 2003
> > [everything associated with speechsc seems to go out late]
> >
> > Since there has been no progress on the protocol analysis
> > document, I have made the following decisions:
> >
> > 1. We need a new protocol.
> >
> > 2. Since no one seems to care, the protocol will be based on
> > BEEP and CORBA.
> >
> > 3. To ensure that people use the protocol, we will publish an
> > ADA binding.
> >
> > Unless I hear otherwise, this is the direction we will go.
> >
> > --
> > - Eric
> >
> >
> > P.S.  If I don't hear otherwise, we can consider the work of
> > the work group complete.
> >
> > _______________________________________________
> > 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

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



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Reply-To: <brian.wyld@eloquant.com>
From: "Brian Wyld" <brian.wyld@eloquant.com>
To: "'Eric Burger'" <eburger@snowshore.com>,
        "'Brian Eberman'" <bse@speechworks.com>,
        "'IETF SPEECHSC (E-mail)'" <speechsc@ietf.org>
Subject: RE: [Speechsc] speechsc as a first-level protocol
Date: Fri, 9 May 2003 17:58:18 +0200
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Sounds good to me IFF we mean by this not neccessarily reinventing the whole
thing, but doing a pick-n-mix from current protocols.....

Brian

 (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, add
MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end context
and move onto thez virgin territory of SV/SI....)




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

> -----Message d'origine-----
> De : speechsc-admin@ietf.org
> [mailto:speechsc-admin@ietf.org]De la part
> de Eric Burger
> Envoyé : Thursday, May 08, 2003 20:21
> À : Brian Eberman; IETF SPEECHSC (E-mail)
> Objet : [Speechsc] speechsc as a first-level protocol
>
>
> Is there consensus on raising speechsc to being a first-level
> protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?
>
> Does anyone strongly object?
>
> Would anyone be really upset if it isn't?
>
> Is there value to it being a first-level protocol inside an
> existing protocol?  That is, use the session establishment
> and proxy infrastructure of SIP or RTSP and have
> speechsc-specific primitives as protocol extensions.
>
> > -----Original Message-----
> > From: Brian Eberman [mailto:bse@speechworks.com]
> > Sent: Mon, May 05, 2003 3:13 PM
> > To: Eric Burger; IETF SPEECHSC (E-mail)
> > Subject: RE: [Speechsc] Administrative Fiat
> >
> >
> > Eric,
> >
> > Seems to be a bit of a consensus here. What are the next steps to be
> > completed?
> >
> > -Brian E.
> >
> > Director Product Management and Services
> > SpeechWorks International
> > 695 Atlantic Avenue
> > Boston, MA 02111
> > 617 428 4444
> >
> >
> > -----Original Message-----
> > From: speechsc-admin@ietf.org
> > [mailto:speechsc-admin@ietf.org]On Behalf
> > Of Rajiv Dharmadhikari
> > Sent: Friday, April 25, 2003 12:51 PM
> > To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
> > Subject: RE: [Speechsc] Administrative Fiat
> >
> >
> > Brian,
> >
> > Can you resend the final draft of the protocl evaluation
> > analysis to the
> > initial team? What is left to be completed? Let's get it out
> > of the way.. I
> > am willing do complete my part since I don't know BEEP, COBOL
> > and CORBA :-)
> >
> > Thanks,
> >
> > -Rajiv
> >
> > -----Original Message-----
> > From: Brian Wyld [mailto:brian.wyld@eloquant.com]
> > Sent: Friday, April 25, 2003 1:46 AM
> > To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
> > Subject: RE: [Speechsc] Administrative Fiat
> >
> >
> > Eric,
> >
> > Sounds good to me...... although I believe there is still a
> > significant base
> > of COBOL applications that may be critical to address.......
> >
> > Or, I contra-propose:
> > 1. the protocol will be RTSP extended to include the current
> > MRCP messages
> > 'natively', with addition of specific  messages for SR/SI,
> > and to define the
> > 'call session' mapping to the MRCP requests.
> >
> > 2. there will be a web services binding....
> >
> > (sorry to be serious here)
> >
> > Brian
> >
> > [Brian Wyld] [brian.wyld@eloquant.com]
> > [Directeur General R&D]
> > [Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
> > [advanced solutions for telecoms and IT services]
> >
> > > -----Message d'origine-----
> > > De : speechsc-admin@ietf.org
> > > [mailto:speechsc-admin@ietf.org]De la part
> > > de Eric Burger
> > > Envoyé : Thursday, April 24, 2003 17:03
> > > À : IETF SPEECHSC (E-mail)
> > > Objet : [Speechsc] Administrative Fiat
> > >
> > >
> > > Date: 1 April 2003
> > > [everything associated with speechsc seems to go out late]
> > >
> > > Since there has been no progress on the protocol analysis
> > > document, I have made the following decisions:
> > >
> > > 1. We need a new protocol.
> > >
> > > 2. Since no one seems to care, the protocol will be based on
> > > BEEP and CORBA.
> > >
> > > 3. To ensure that people use the protocol, we will publish an
> > > ADA binding.
> > >
> > > Unless I hear otherwise, this is the direction we will go.
> > >
> > > --
> > > - Eric
> > >
> > >
> > > P.S.  If I don't hear otherwise, we can consider the work of
> > > the work group complete.
> > >
> > > _______________________________________________
> > > 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
>

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



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Message-ID: <3EBBDC09.2050406@cisco.com>
Date: Fri, 09 May 2003 09:49:13 -0700
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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        "'Brian Eberman'"
 <bse@speechworks.com>,
        "'IETF SPEECHSC (E-mail)'" <speechsc@ietf.org>
Subject: Re: [Speechsc] speechsc as a first-level protocol
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--------------060902010006030401020803
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Hi Brian,
       Great. I am half way through a proposal trying to do exactly=20
that. Since there was not much activity, I thought a proposal document=20
would drive some discussion. So after consulting with Eric, I have been=20
writing a proposal, doing exactly that.

As part 1, I have taken out all RTSP tunnelling and related aspects from=20
the currentt MRCP specification, this is done.  As Part to 2, I am=20
adding/advocating SIP based INVITE/BYE extensions a choice to create and=20
manage the session.

I will hopefully have this proposal done by the beginning of the week=20
after next.

Sarvi


Brian Wyld wrote:

>Sounds good to me IFF we mean by this not neccessarily reinventing the w=
hole
>thing, but doing a pick-n-mix from current protocols.....
>
>Brian
>
> (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, a=
dd
>MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end cont=
ext
>and move onto thez virgin territory of SV/SI....)
>
>
>
>
>[Brian Wyld] [brian.wyld@eloquant.com]
>[Directeur General R&D]
>[Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
>[advanced solutions for telecoms and IT services]
>
> =20
>
>>-----Message d'origine-----
>>De : speechsc-admin@ietf.org
>>[mailto:speechsc-admin@ietf.org]De la part
>>de Eric Burger
>>Envoy=E9 : Thursday, May 08, 2003 20:21
>>=C0 : Brian Eberman; IETF SPEECHSC (E-mail)
>>Objet : [Speechsc] speechsc as a first-level protocol
>>
>>
>>Is there consensus on raising speechsc to being a first-level
>>protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?
>>
>>Does anyone strongly object?
>>
>>Would anyone be really upset if it isn't?
>>
>>Is there value to it being a first-level protocol inside an
>>existing protocol?  That is, use the session establishment
>>and proxy infrastructure of SIP or RTSP and have
>>speechsc-specific primitives as protocol extensions.
>>
>>   =20
>>
>>>-----Original Message-----
>>>From: Brian Eberman [mailto:bse@speechworks.com]
>>>Sent: Mon, May 05, 2003 3:13 PM
>>>To: Eric Burger; IETF SPEECHSC (E-mail)
>>>Subject: RE: [Speechsc] Administrative Fiat
>>>
>>>
>>>Eric,
>>>
>>>Seems to be a bit of a consensus here. What are the next steps to be
>>>completed?
>>>
>>>-Brian E.
>>>
>>>Director Product Management and Services
>>>SpeechWorks International
>>>695 Atlantic Avenue
>>>Boston, MA 02111
>>>617 428 4444
>>>
>>>
>>>-----Original Message-----
>>>From: speechsc-admin@ietf.org
>>>[mailto:speechsc-admin@ietf.org]On Behalf
>>>Of Rajiv Dharmadhikari
>>>Sent: Friday, April 25, 2003 12:51 PM
>>>To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
>>>Subject: RE: [Speechsc] Administrative Fiat
>>>
>>>
>>>Brian,
>>>
>>>Can you resend the final draft of the protocl evaluation
>>>analysis to the
>>>initial team? What is left to be completed? Let's get it out
>>>of the way.. I
>>>am willing do complete my part since I don't know BEEP, COBOL
>>>and CORBA :-)
>>>
>>>Thanks,
>>>
>>>-Rajiv
>>>
>>>-----Original Message-----
>>>From: Brian Wyld [mailto:brian.wyld@eloquant.com]
>>>Sent: Friday, April 25, 2003 1:46 AM
>>>To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
>>>Subject: RE: [Speechsc] Administrative Fiat
>>>
>>>
>>>Eric,
>>>
>>>Sounds good to me...... although I believe there is still a
>>>significant base
>>>of COBOL applications that may be critical to address.......
>>>
>>>Or, I contra-propose:
>>>1. the protocol will be RTSP extended to include the current
>>>MRCP messages
>>>'natively', with addition of specific  messages for SR/SI,
>>>and to define the
>>>'call session' mapping to the MRCP requests.
>>>
>>>2. there will be a web services binding....
>>>
>>>(sorry to be serious here)
>>>
>>>Brian
>>>
>>>[Brian Wyld] [brian.wyld@eloquant.com]
>>>[Directeur General R&D]
>>>[Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
>>>[advanced solutions for telecoms and IT services]
>>>
>>>     =20
>>>
>>>>-----Message d'origine-----
>>>>De : speechsc-admin@ietf.org
>>>>[mailto:speechsc-admin@ietf.org]De la part
>>>>de Eric Burger
>>>>Envoy=E9 : Thursday, April 24, 2003 17:03
>>>>=C0 : IETF SPEECHSC (E-mail)
>>>>Objet : [Speechsc] Administrative Fiat
>>>>
>>>>
>>>>Date: 1 April 2003
>>>>[everything associated with speechsc seems to go out late]
>>>>
>>>>Since there has been no progress on the protocol analysis
>>>>document, I have made the following decisions:
>>>>
>>>>1. We need a new protocol.
>>>>
>>>>2. Since no one seems to care, the protocol will be based on
>>>>BEEP and CORBA.
>>>>
>>>>3. To ensure that people use the protocol, we will publish an
>>>>ADA binding.
>>>>
>>>>Unless I hear otherwise, this is the direction we will go.
>>>>
>>>>--
>>>>- Eric
>>>>
>>>>
>>>>P.S.  If I don't hear otherwise, we can consider the work of
>>>>the work group complete.
>>>>
>>>>_______________________________________________
>>>>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
>>>_______________________________________________
>>>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
>


--------------060902010006030401020803
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<head>
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<body>
Hi Brian,<br>
&nbsp; &nbsp; &nbsp; &nbsp;Great. I am half way through a proposal trying to do exactly that.
Since there was not much activity, I thought a proposal document would drive
some discussion. So after consulting with Eric, I have been writing a proposal,
doing exactly that.<br>
<br>
As part 1, I have taken out all RTSP tunnelling and related aspects from
the currentt MRCP specification, this is done.&nbsp; As Part to 2, I am adding/advocating
SIP based INVITE/BYE extensions a choice to create and manage the session.<br>
<br>
I will hopefully have this proposal done by the beginning of the week after
next.<br>
<br>
Sarvi<br>
<br>
<br>
Brian Wyld wrote:<br>
<blockquote type="cite"
 cite="mid14cd01c31643$cf3d8040$8300010a@hq.eloquant.com">
  <pre wrap="">Sounds good to me IFF we mean by this not neccessarily reinventing the whole
thing, but doing a pick-n-mix from current protocols.....

Brian

 (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, add
MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end context
and move onto thez virgin territory of SV/SI....)




[Brian Wyld] [<a class="moz-txt-link-abbreviated" href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a>]
[Directeur General R&amp;D]
[Eloquant SA] [+33 476 77 46 92] [<a class="moz-txt-link-abbreviated" href="http://www.eloquant.com">www.eloquant.com</a>]
[advanced solutions for telecoms and IT services]

  </pre>
  <blockquote type="cite">
    <pre wrap="">-----Message d'origine-----
De : <a class="moz-txt-link-abbreviated" href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>]De la part
de Eric Burger
Envoy&eacute; : Thursday, May 08, 2003 20:21
&Agrave; : Brian Eberman; IETF SPEECHSC (E-mail)
Objet : [Speechsc] speechsc as a first-level protocol


Is there consensus on raising speechsc to being a first-level
protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?

Does anyone strongly object?

Would anyone be really upset if it isn't?

Is there value to it being a first-level protocol inside an
existing protocol?  That is, use the session establishment
and proxy infrastructure of SIP or RTSP and have
speechsc-specific primitives as protocol extensions.

    </pre>
    <blockquote type="cite">
      <pre wrap="">-----Original Message-----
From: Brian Eberman [<a class="moz-txt-link-freetext" href="mailto:bse@speechworks.com">mailto:bse@speechworks.com</a>]
Sent: Mon, May 05, 2003 3:13 PM
To: Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Seems to be a bit of a consensus here. What are the next steps to be
completed?

-Brian E.

Director Product Management and Services
SpeechWorks International
695 Atlantic Avenue
Boston, MA 02111
617 428 4444


-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>]On Behalf
Of Rajiv Dharmadhikari
Sent: Friday, April 25, 2003 12:51 PM
To: <a class="moz-txt-link-abbreviated" href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a>; Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Brian,

Can you resend the final draft of the protocl evaluation
analysis to the
initial team? What is left to be completed? Let's get it out
of the way.. I
am willing do complete my part since I don't know BEEP, COBOL
and CORBA :-)

Thanks,

-Rajiv

-----Original Message-----
From: Brian Wyld [<a class="moz-txt-link-freetext" href="mailto:brian.wyld@eloquant.com">mailto:brian.wyld@eloquant.com</a>]
Sent: Friday, April 25, 2003 1:46 AM
To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Sounds good to me...... although I believe there is still a
significant base
of COBOL applications that may be critical to address.......

Or, I contra-propose:
1. the protocol will be RTSP extended to include the current
MRCP messages
'natively', with addition of specific  messages for SR/SI,
and to define the
'call session' mapping to the MRCP requests.

2. there will be a web services binding....

(sorry to be serious here)

Brian

[Brian Wyld] [<a class="moz-txt-link-abbreviated" href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a>]
[Directeur General R&amp;D]
[Eloquant SA] [+33 476 77 46 92] [<a class="moz-txt-link-abbreviated" href="http://www.eloquant.com">www.eloquant.com</a>]
[advanced solutions for telecoms and IT services]

      </pre>
      <blockquote type="cite">
        <pre wrap="">-----Message d'origine-----
De : <a class="moz-txt-link-abbreviated" href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>]De la part
de Eric Burger
Envoy&eacute; : Thursday, April 24, 2003 17:03
&Agrave; : IETF SPEECHSC (E-mail)
Objet : [Speechsc] Administrative Fiat


Date: 1 April 2003
[everything associated with speechsc seems to go out late]

Since there has been no progress on the protocol analysis
document, I have made the following decisions:

1. We need a new protocol.

2. Since no one seems to care, the protocol will be based on
BEEP and CORBA.

3. To ensure that people use the protocol, we will publish an
ADA binding.

Unless I hear otherwise, this is the direction we will go.

--
- Eric


P.S.  If I don't hear otherwise, we can consider the work of
the work group complete.

_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

        </pre>
      </blockquote>
      <pre wrap="">_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>
_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

      </pre>
    </blockquote>
    <pre wrap="">_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

    </pre>
  </blockquote>
  <pre wrap=""><!---->
_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

  </pre>
</blockquote>
<br>
</body>
</html>

--------------060902010006030401020803--

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



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Reply-To: <brian.wyld@eloquant.com>
From: "Brian Wyld" <brian.wyld@eloquant.com>
To: "'Sarvi Shanmugham'" <sarvi@cisco.com>
Cc: "'Eric Burger'" <eburger@snowshore.com>,
        "'Brian Eberman'" <bse@speechworks.com>,
        "'IETF SPEECHSC (E-mail)'" <speechsc@ietf.org>
Subject: RE: [Speechsc] speechsc as a first-level protocol
Date: Fri, 9 May 2003 18:51:25 +0200
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Hi,

Great! However, I would REALLY like to see separation of the media
setup/teardown function and the concept of 'mrcp' session (which could be
mapped onto a call nicely).

My experience with MRCP so far is that this can be a prickly issue (for
those ASR vendors who have strong licensing view about ports and calls!!!=
)

Brian



  -----Message d'origine-----
  De : Sarvi Shanmugham [mailto:sarvi@cisco.com]
  Envoy=E9 : Friday, May 09, 2003 18:49
  =C0 : brian.wyld@eloquant.com
  Cc : 'Eric Burger'; 'Brian Eberman'; 'IETF SPEECHSC (E-mail)'
  Objet : Re: [Speechsc] speechsc as a first-level protocol


  Hi Brian,
         Great. I am half way through a proposal trying to do exactly tha=
t.
Since there was not much activity, I thought a proposal document would dr=
ive
some discussion. So after consulting with Eric, I have been writing a
proposal, doing exactly that.

  As part 1, I have taken out all RTSP tunnelling and related aspects fro=
m
the currentt MRCP specification, this is done.  As Part to 2, I am
adding/advocating SIP based INVITE/BYE extensions a choice to create and
manage the session.

  I will hopefully have this proposal done by the beginning of the week
after next.

  Sarvi


  Brian Wyld wrote:

Sounds good to me IFF we mean by this not neccessarily reinventing the wh=
ole
thing, but doing a pick-n-mix from current protocols.....

Brian

 (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, ad=
d
MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end conte=
xt
and move onto thez virgin territory of SV/SI....)




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


-----Message d'origine-----
De : speechsc-admin@ietf.org
[mailto:speechsc-admin@ietf.org]De la part
de Eric Burger
Envoy=E9 : Thursday, May 08, 2003 20:21
=C0 : Brian Eberman; IETF SPEECHSC (E-mail)
Objet : [Speechsc] speechsc as a first-level protocol


Is there consensus on raising speechsc to being a first-level
protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?

Does anyone strongly object?

Would anyone be really upset if it isn't?

Is there value to it being a first-level protocol inside an
existing protocol?  That is, use the session establishment
and proxy infrastructure of SIP or RTSP and have
speechsc-specific primitives as protocol extensions.


-----Original Message-----
From: Brian Eberman [mailto:bse@speechworks.com]
Sent: Mon, May 05, 2003 3:13 PM
To: Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Seems to be a bit of a consensus here. What are the next steps to be
completed?

-Brian E.

Director Product Management and Services
SpeechWorks International
695 Atlantic Avenue
Boston, MA 02111
617 428 4444


-----Original Message-----
From: speechsc-admin@ietf.org
[mailto:speechsc-admin@ietf.org]On Behalf
Of Rajiv Dharmadhikari
Sent: Friday, April 25, 2003 12:51 PM
To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Brian,

Can you resend the final draft of the protocl evaluation
analysis to the
initial team? What is left to be completed? Let's get it out
of the way.. I
am willing do complete my part since I don't know BEEP, COBOL
and CORBA :-)

Thanks,

-Rajiv

-----Original Message-----
From: Brian Wyld [mailto:brian.wyld@eloquant.com]
Sent: Friday, April 25, 2003 1:46 AM
To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Sounds good to me...... although I believe there is still a
significant base
of COBOL applications that may be critical to address.......

Or, I contra-propose:
1. the protocol will be RTSP extended to include the current
MRCP messages
'natively', with addition of specific  messages for SR/SI,
and to define the
'call session' mapping to the MRCP requests.

2. there will be a web services binding....

(sorry to be serious here)

Brian

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


-----Message d'origine-----
De : speechsc-admin@ietf.org
[mailto:speechsc-admin@ietf.org]De la part
de Eric Burger
Envoy=E9 : Thursday, April 24, 2003 17:03
=C0 : IETF SPEECHSC (E-mail)
Objet : [Speechsc] Administrative Fiat


Date: 1 April 2003
[everything associated with speechsc seems to go out late]

Since there has been no progress on the protocol analysis
document, I have made the following decisions:

1. We need a new protocol.

2. Since no one seems to care, the protocol will be based on
BEEP and CORBA.

3. To ensure that people use the protocol, we will publish an
ADA binding.

Unless I hear otherwise, this is the direction we will go.

--
- Eric


P.S.  If I don't hear otherwise, we can consider the work of
the work group complete.

_______________________________________________
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



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





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	charset="iso-8859-1"
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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE></TITLE>

<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D366414916-09052003>Hi,</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D366414916-09052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D366414916-09052003>Great!=20
However, I would REALLY like to see separation of the media =
setup/teardown=20
function and the concept of 'mrcp' session (which could be mapped onto a =
call=20
nicely). </SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D366414916-09052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D366414916-09052003>My=20
experience with MRCP so far is that this can be a prickly issue (for =
those ASR=20
vendors who have strong licensing view about ports and=20
calls!!!)</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D366414916-09052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D366414916-09052003>Brian</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<P><FONT size=3D2>&nbsp;</FONT> </P>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: =
5px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Message d'origine-----<BR><B>De&nbsp;:</B> Sarvi =
Shanmugham=20
  [mailto:sarvi@cisco.com]<BR><B>Envoy=E9&nbsp;:</B> Friday, May 09, =
2003=20
  18:49<BR><B>=C0&nbsp;:</B> brian.wyld@eloquant.com<BR><B>Cc&nbsp;:</B> =
'Eric=20
  Burger'; 'Brian Eberman'; 'IETF SPEECHSC =
(E-mail)'<BR><B>Objet&nbsp;:</B> Re:=20
  [Speechsc] speechsc as a first-level protocol<BR><BR></DIV></FONT>Hi=20
  Brian,<BR>&nbsp; &nbsp; &nbsp; &nbsp;Great. I am half way through a =
proposal=20
  trying to do exactly that. Since there was not much activity, I =
thought a=20
  proposal document would drive some discussion. So after consulting =
with Eric,=20
  I have been writing a proposal, doing exactly that.<BR><BR>As part 1, =
I have=20
  taken out all RTSP tunnelling and related aspects from the currentt =
MRCP=20
  specification, this is done.&nbsp; As Part to 2, I am =
adding/advocating SIP=20
  based INVITE/BYE extensions a choice to create and manage the=20
  session.<BR><BR>I will hopefully have this proposal done by the =
beginning of=20
  the week after next.<BR><BR>Sarvi<BR><BR><BR>Brian Wyld wrote:<BR>
  <BLOCKQUOTE cite=3D"mid14cd01c31643$cf3d8040$8300010a@hq.eloquant.com" =

  type=3D"cite"><PRE wrap=3D"">Sounds good to me IFF we mean by this not =
neccessarily reinventing the whole
thing, but doing a pick-n-mix from current protocols.....

Brian

 (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, =
add
MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end =
context
and move onto thez virgin territory of SV/SI....)




[Brian Wyld] [<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</A>]
[Directeur General R&amp;D]
[Eloquant SA] [+33 476 77 46 92] [<A class=3Dmoz-txt-link-abbreviated =
href=3D"http://www.eloquant.com">www.eloquant.com</A>]
[advanced solutions for telecoms and IT services]

  </PRE>
    <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">-----Message d'origine-----
De : <A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</A>
[<A class=3Dmoz-txt-link-freetext =
href=3D"mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</A=
>]De la part
de Eric Burger
Envoy=E9 : Thursday, May 08, 2003 20:21
=C0 : Brian Eberman; IETF SPEECHSC (E-mail)
Objet : [Speechsc] speechsc as a first-level protocol


Is there consensus on raising speechsc to being a first-level
protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?

Does anyone strongly object?

Would anyone be really upset if it isn't?

Is there value to it being a first-level protocol inside an
existing protocol?  That is, use the session establishment
and proxy infrastructure of SIP or RTSP and have
speechsc-specific primitives as protocol extensions.

    </PRE>
      <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">-----Original =
Message-----
From: Brian Eberman [<A class=3Dmoz-txt-link-freetext =
href=3D"mailto:bse@speechworks.com">mailto:bse@speechworks.com</A>]
Sent: Mon, May 05, 2003 3:13 PM
To: Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Seems to be a bit of a consensus here. What are the next steps to be
completed?

-Brian E.

Director Product Management and Services
SpeechWorks International
695 Atlantic Avenue
Boston, MA 02111
617 428 4444


-----Original Message-----
From: <A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</A>
[<A class=3Dmoz-txt-link-freetext =
href=3D"mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</A=
>]On Behalf
Of Rajiv Dharmadhikari
Sent: Friday, April 25, 2003 12:51 PM
To: <A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</A>; =
Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Brian,

Can you resend the final draft of the protocl evaluation
analysis to the
initial team? What is left to be completed? Let's get it out
of the way.. I
am willing do complete my part since I don't know BEEP, COBOL
and CORBA :-)

Thanks,

-Rajiv

-----Original Message-----
From: Brian Wyld [<A class=3Dmoz-txt-link-freetext =
href=3D"mailto:brian.wyld@eloquant.com">mailto:brian.wyld@eloquant.com</A=
>]
Sent: Friday, April 25, 2003 1:46 AM
To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Sounds good to me...... although I believe there is still a
significant base
of COBOL applications that may be critical to address.......

Or, I contra-propose:
1. the protocol will be RTSP extended to include the current
MRCP messages
'natively', with addition of specific  messages for SR/SI,
and to define the
'call session' mapping to the MRCP requests.

2. there will be a web services binding....

(sorry to be serious here)

Brian

[Brian Wyld] [<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</A>]
[Directeur General R&amp;D]
[Eloquant SA] [+33 476 77 46 92] [<A class=3Dmoz-txt-link-abbreviated =
href=3D"http://www.eloquant.com">www.eloquant.com</A>]
[advanced solutions for telecoms and IT services]

      </PRE>
        <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">-----Message =
d'origine-----
De : <A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</A>
[<A class=3Dmoz-txt-link-freetext =
href=3D"mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</A=
>]De la part
de Eric Burger
Envoy=E9 : Thursday, April 24, 2003 17:03
=C0 : IETF SPEECHSC (E-mail)
Objet : [Speechsc] Administrative Fiat


Date: 1 April 2003
[everything associated with speechsc seems to go out late]

Since there has been no progress on the protocol analysis
document, I have made the following decisions:

1. We need a new protocol.

2. Since no one seems to care, the protocol will be based on
BEEP and CORBA.

3. To ensure that people use the protocol, we will publish an
ADA binding.

Unless I hear otherwise, this is the direction we will go.

--
- Eric


P.S.  If I don't hear otherwise, we can consider the work of
the work group complete.

_______________________________________________
Speechsc mailing list
<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:Speechsc@ietf.org">Speechsc@ietf.org</A>
<A class=3Dmoz-txt-link-freetext =
href=3D"https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.iet=
f.org/mailman/listinfo/speechsc</A>

        </PRE></BLOCKQUOTE><PRE =
wrap=3D"">_______________________________________________
Speechsc mailing list
<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:Speechsc@ietf.org">Speechsc@ietf.org</A>
<A class=3Dmoz-txt-link-freetext =
href=3D"https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.iet=
f.org/mailman/listinfo/speechsc</A>
_______________________________________________
Speechsc mailing list
<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:Speechsc@ietf.org">Speechsc@ietf.org</A>
<A class=3Dmoz-txt-link-freetext =
href=3D"https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.iet=
f.org/mailman/listinfo/speechsc</A>

      </PRE></BLOCKQUOTE><PRE =
wrap=3D"">_______________________________________________
Speechsc mailing list
<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:Speechsc@ietf.org">Speechsc@ietf.org</A>
<A class=3Dmoz-txt-link-freetext =
href=3D"https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.iet=
f.org/mailman/listinfo/speechsc</A>

    </PRE></BLOCKQUOTE><PRE wrap=3D""><!---->
_______________________________________________
Speechsc mailing list
<A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:Speechsc@ietf.org">Speechsc@ietf.org</A>
<A class=3Dmoz-txt-link-freetext =
href=3D"https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.iet=
f.org/mailman/listinfo/speechsc</A>

  </PRE></BLOCKQUOTE><BR></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_14E9_01C3165B.FE974F70--

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



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Message-ID: <3EBBE11D.7000605@cisco.com>
Date: Fri, 09 May 2003 10:10:53 -0700
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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Subject: Re: [Speechsc] speechsc as a first-level protocol
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--------------080602050605060609010403
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Absolutely. This will only be an initial proposal.
We need to work out ways of differentiating/separating  the session to=20
the the Media server and the call that is being treated and figure how=20
to share the media  session and use it for different calls.

Sarvi

Brian Wyld wrote:

> Hi,
> =20
> Great! However, I would REALLY like to see separation of the media=20
> setup/teardown function and the concept of 'mrcp' session (which could=20
> be mapped onto a call nicely).
> =20
> My experience with MRCP so far is that this can be a prickly issue=20
> (for those ASR vendors who have strong licensing view about ports and=20
> calls!!!)
> =20
> Brian
> =20
>
> =20
>
>     -----Message d'origine-----
>     De : Sarvi Shanmugham [mailto:sarvi@cisco.com]
>     Envoy=E9 : Friday, May 09, 2003 18:49
>     =C0 : brian.wyld@eloquant.com
>     Cc : 'Eric Burger'; 'Brian Eberman'; 'IETF SPEECHSC (E-mail)'
>     Objet : Re: [Speechsc] speechsc as a first-level protocol
>
>     Hi Brian,
>            Great. I am half way through a proposal trying to do
>     exactly that. Since there was not much activity, I thought a
>     proposal document would drive some discussion. So after consulting
>     with Eric, I have been writing a proposal, doing exactly that.
>
>     As part 1, I have taken out all RTSP tunnelling and related
>     aspects from the currentt MRCP specification, this is done.  As
>     Part to 2, I am adding/advocating SIP based INVITE/BYE extensions
>     a choice to create and manage the session.
>
>     I will hopefully have this proposal done by the beginning of the
>     week after next.
>
>     Sarvi
>
>
>     Brian Wyld wrote:
>
>>Sounds good to me IFF we mean by this not neccessarily reinventing the =
whole
>>thing, but doing a pick-n-mix from current protocols.....
>>
>>Brian
>>
>> (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, =
add
>>MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end con=
text
>>and move onto thez virgin territory of SV/SI....)
>>
>>
>>
>>
>>[Brian Wyld] [brian.wyld@eloquant.com]
>>[Directeur General R&D]
>>[Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
>>[advanced solutions for telecoms and IT services]
>>
>> =20
>>
>>>-----Message d'origine-----
>>>De : speechsc-admin@ietf.org
>>>[mailto:speechsc-admin@ietf.org]De la part
>>>de Eric Burger
>>>Envoy=E9 : Thursday, May 08, 2003 20:21
>>>=C0 : Brian Eberman; IETF SPEECHSC (E-mail)
>>>Objet : [Speechsc] speechsc as a first-level protocol
>>>
>>>
>>>Is there consensus on raising speechsc to being a first-level
>>>protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?
>>>
>>>Does anyone strongly object?
>>>
>>>Would anyone be really upset if it isn't?
>>>
>>>Is there value to it being a first-level protocol inside an
>>>existing protocol?  That is, use the session establishment
>>>and proxy infrastructure of SIP or RTSP and have
>>>speechsc-specific primitives as protocol extensions.
>>>
>>>   =20
>>>
>>>>-----Original Message-----
>>>>From: Brian Eberman [mailto:bse@speechworks.com]
>>>>Sent: Mon, May 05, 2003 3:13 PM
>>>>To: Eric Burger; IETF SPEECHSC (E-mail)
>>>>Subject: RE: [Speechsc] Administrative Fiat
>>>>
>>>>
>>>>Eric,
>>>>
>>>>Seems to be a bit of a consensus here. What are the next steps to be
>>>>completed?
>>>>
>>>>-Brian E.
>>>>
>>>>Director Product Management and Services
>>>>SpeechWorks International
>>>>695 Atlantic Avenue
>>>>Boston, MA 02111
>>>>617 428 4444
>>>>
>>>>
>>>>-----Original Message-----
>>>>From: speechsc-admin@ietf.org
>>>>[mailto:speechsc-admin@ietf.org]On Behalf
>>>>Of Rajiv Dharmadhikari
>>>>Sent: Friday, April 25, 2003 12:51 PM
>>>>To: brian.wyld@eloquant.com; Eric Burger; IETF SPEECHSC (E-mail)
>>>>Subject: RE: [Speechsc] Administrative Fiat
>>>>
>>>>
>>>>Brian,
>>>>
>>>>Can you resend the final draft of the protocl evaluation
>>>>analysis to the
>>>>initial team? What is left to be completed? Let's get it out
>>>>of the way.. I
>>>>am willing do complete my part since I don't know BEEP, COBOL
>>>>and CORBA :-)
>>>>
>>>>Thanks,
>>>>
>>>>-Rajiv
>>>>
>>>>-----Original Message-----
>>>>From: Brian Wyld [mailto:brian.wyld@eloquant.com]
>>>>Sent: Friday, April 25, 2003 1:46 AM
>>>>To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
>>>>Subject: RE: [Speechsc] Administrative Fiat
>>>>
>>>>
>>>>Eric,
>>>>
>>>>Sounds good to me...... although I believe there is still a
>>>>significant base
>>>>of COBOL applications that may be critical to address.......
>>>>
>>>>Or, I contra-propose:
>>>>1. the protocol will be RTSP extended to include the current
>>>>MRCP messages
>>>>'natively', with addition of specific  messages for SR/SI,
>>>>and to define the
>>>>'call session' mapping to the MRCP requests.
>>>>
>>>>2. there will be a web services binding....
>>>>
>>>>(sorry to be serious here)
>>>>
>>>>Brian
>>>>
>>>>[Brian Wyld] [brian.wyld@eloquant.com]
>>>>[Directeur General R&D]
>>>>[Eloquant SA] [+33 476 77 46 92] [www.eloquant.com]
>>>>[advanced solutions for telecoms and IT services]
>>>>
>>>>     =20
>>>>
>>>>>-----Message d'origine-----
>>>>>De : speechsc-admin@ietf.org
>>>>>[mailto:speechsc-admin@ietf.org]De la part
>>>>>de Eric Burger
>>>>>Envoy=E9 : Thursday, April 24, 2003 17:03
>>>>>=C0 : IETF SPEECHSC (E-mail)
>>>>>Objet : [Speechsc] Administrative Fiat
>>>>>
>>>>>
>>>>>Date: 1 April 2003
>>>>>[everything associated with speechsc seems to go out late]
>>>>>
>>>>>Since there has been no progress on the protocol analysis
>>>>>document, I have made the following decisions:
>>>>>
>>>>>1. We need a new protocol.
>>>>>
>>>>>2. Since no one seems to care, the protocol will be based on
>>>>>BEEP and CORBA.
>>>>>
>>>>>3. To ensure that people use the protocol, we will publish an
>>>>>ADA binding.
>>>>>
>>>>>Unless I hear otherwise, this is the direction we will go.
>>>>>
>>>>>--
>>>>>- Eric
>>>>>
>>>>>
>>>>>P.S.  If I don't hear otherwise, we can consider the work of
>>>>>the work group complete.
>>>>>
>>>>>_______________________________________________
>>>>>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
>>>>_______________________________________________
>>>>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
>>
>


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<head>
  <meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
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<body>
Absolutely. This will only be an initial proposal. <br>
We need to work out ways of differentiating/separating&nbsp; the session to the
the Media server and the call that is being treated and figure how to share
the media &nbsp;session and use it for different calls.<br>
<br>
Sarvi<br>
<br>
Brian Wyld wrote:<br>
<blockquote type="cite"
 cite="mid14e801c3164b$3b0e7f70$8300010a@hq.eloquant.com">  
  <meta http-equiv="Content-Type" content="text/html; ">
  <title></title>
   
  <meta content="MSHTML 5.00.2920.0" name="GENERATOR">
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003">Hi,</span></font></div>
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003"></span></font>&nbsp;</div>
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003">Great!  However, I would REALLY like to see separation
of the media setup/teardown  function and the concept of 'mrcp' session (which
could be mapped onto a call  nicely). </span></font></div>
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003"></span></font>&nbsp;</div>
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003">My  experience with MRCP so far is that this
can be a prickly issue (for those ASR  vendors who have strong licensing
view about ports and  calls!!!)</span></font></div>
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003"></span></font>&nbsp;</div>
 
  <div><font color="#0000ff" face="Arial" size="2"><span
 class="366414916-09052003">Brian</span></font></div>
 
  <div>&nbsp;</div>
 
  <p><font size="2">&nbsp;</font> </p>
 
  <blockquote
 style="border-left: 2px solid rgb(0,0,255); margin-left: 5px; padding-left: 5px;"> 
  
    <div align="left" class="OutlookMessageHeader" dir="ltr"><font
 face="Tahoma" size="2">-----Message d'origine-----<br>
    <b>De&nbsp;:</b> Sarvi Shanmugham    [<a class="moz-txt-link-freetext" href="mailto:sarvi@cisco.com">mailto:sarvi@cisco.com</a>]<br>
    <b>Envoy&eacute;&nbsp;:</b> Friday, May 09, 2003    18:49<br>
    <b>&Agrave;&nbsp;:</b> <a class="moz-txt-link-abbreviated" href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a><br>
    <b>Cc&nbsp;:</b> 'Eric    Burger'; 'Brian Eberman'; 'IETF SPEECHSC (E-mail)'<br>
    <b>Objet&nbsp;:</b> Re:    [Speechsc] speechsc as a first-level protocol<br>
    <br>
    </font></div>
Hi    Brian,<br>
&nbsp; &nbsp; &nbsp; &nbsp;Great. I am half way through a proposal    trying to do exactly that.
Since there was not much activity, I thought a    proposal document would
drive some discussion. So after consulting with Eric,    I have been writing
a proposal, doing exactly that.<br>
    <br>
As part 1, I have    taken out all RTSP tunnelling and related aspects from
the currentt MRCP    specification, this is done.&nbsp; As Part to 2, I am adding/advocating
SIP    based INVITE/BYE extensions a choice to create and manage the    session.<br>
    <br>
I will hopefully have this proposal done by the beginning of    the week
after next.<br>
    <br>
Sarvi<br>
    <br>
    <br>
Brian Wyld wrote:<br>
   
    <blockquote cite="mid14cd01c31643$cf3d8040$8300010a@hq.eloquant.com"
 type="cite">
      <pre wrap="">Sounds good to me IFF we mean by this not neccessarily reinventing the whole
thing, but doing a pick-n-mix from current protocols.....

Brian

 (eg, stick the SETUP and TEARDOWN messages from RTSP into MRCP draft, add
MRCPSESSIONSTART/MRCPSESSIONEND messages to identify call start/end context
and move onto thez virgin territory of SV/SI....)




[Brian Wyld] [<a class="moz-txt-link-abbreviated"
 href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a>]
[Directeur General R&amp;D]
[Eloquant SA] [+33 476 77 46 92] [<a class="moz-txt-link-abbreviated"
 href="http://www.eloquant.com">www.eloquant.com</a>]
[advanced solutions for telecoms and IT services]

  </pre>
     
      <blockquote type="cite">
        <pre wrap="">-----Message d'origine-----
De : <a class="moz-txt-link-abbreviated"
 href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>]De la part
de Eric Burger
Envoy&eacute; : Thursday, May 08, 2003 20:21
&Agrave; : Brian Eberman; IETF SPEECHSC (E-mail)
Objet : [Speechsc] speechsc as a first-level protocol


Is there consensus on raising speechsc to being a first-level
protocol?  That is, it will NOT be embedded in SIP, RTSP, or :-) SNMP?

Does anyone strongly object?

Would anyone be really upset if it isn't?

Is there value to it being a first-level protocol inside an
existing protocol?  That is, use the session establishment
and proxy infrastructure of SIP or RTSP and have
speechsc-specific primitives as protocol extensions.

    </pre>
       
        <blockquote type="cite">
          <pre wrap="">-----Original Message-----
From: Brian Eberman [<a class="moz-txt-link-freetext"
 href="mailto:bse@speechworks.com">mailto:bse@speechworks.com</a>]
Sent: Mon, May 05, 2003 3:13 PM
To: Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Seems to be a bit of a consensus here. What are the next steps to be
completed?

-Brian E.

Director Product Management and Services
SpeechWorks International
695 Atlantic Avenue
Boston, MA 02111
617 428 4444


-----Original Message-----
From: <a class="moz-txt-link-abbreviated"
 href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>]On Behalf
Of Rajiv Dharmadhikari
Sent: Friday, April 25, 2003 12:51 PM
To: <a class="moz-txt-link-abbreviated"
 href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a>; Eric Burger; IETF SPEECHSC (E-mail)
Subject: RE: [Speechsc] Administrative Fiat


Brian,

Can you resend the final draft of the protocl evaluation
analysis to the
initial team? What is left to be completed? Let's get it out
of the way.. I
am willing do complete my part since I don't know BEEP, COBOL
and CORBA :-)

Thanks,

-Rajiv

-----Original Message-----
From: Brian Wyld [<a class="moz-txt-link-freetext"
 href="mailto:brian.wyld@eloquant.com">mailto:brian.wyld@eloquant.com</a>]
Sent: Friday, April 25, 2003 1:46 AM
To: 'Eric Burger'; 'IETF SPEECHSC (E-mail)'
Subject: RE: [Speechsc] Administrative Fiat


Eric,

Sounds good to me...... although I believe there is still a
significant base
of COBOL applications that may be critical to address.......

Or, I contra-propose:
1. the protocol will be RTSP extended to include the current
MRCP messages
'natively', with addition of specific  messages for SR/SI,
and to define the
'call session' mapping to the MRCP requests.

2. there will be a web services binding....

(sorry to be serious here)

Brian

[Brian Wyld] [<a class="moz-txt-link-abbreviated"
 href="mailto:brian.wyld@eloquant.com">brian.wyld@eloquant.com</a>]
[Directeur General R&amp;D]
[Eloquant SA] [+33 476 77 46 92] [<a class="moz-txt-link-abbreviated"
 href="http://www.eloquant.com">www.eloquant.com</a>]
[advanced solutions for telecoms and IT services]

      </pre>
         
          <blockquote type="cite">
            <pre wrap="">-----Message d'origine-----
De : <a class="moz-txt-link-abbreviated"
 href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>]De la part
de Eric Burger
Envoy&eacute; : Thursday, April 24, 2003 17:03
&Agrave; : IETF SPEECHSC (E-mail)
Objet : [Speechsc] Administrative Fiat


Date: 1 April 2003
[everything associated with speechsc seems to go out late]

Since there has been no progress on the protocol analysis
document, I have made the following decisions:

1. We need a new protocol.

2. Since no one seems to care, the protocol will be based on
BEEP and CORBA.

3. To ensure that people use the protocol, we will publish an
ADA binding.

Unless I hear otherwise, this is the direction we will go.

--
- Eric


P.S.  If I don't hear otherwise, we can consider the work of
the work group complete.

_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext"
 href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

        </pre>
          </blockquote>
          <pre wrap="">_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext"
 href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>
_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext"
 href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

      </pre>
        </blockquote>
        <pre wrap="">_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext"
 href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

    </pre>
      </blockquote>
      <pre wrap=""><!---->
_______________________________________________
Speechsc mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Speechsc@ietf.org">Speechsc@ietf.org</a>
<a class="moz-txt-link-freetext"
 href="https://www1.ietf.org/mailman/listinfo/speechsc">https://www1.ietf.org/mailman/listinfo/speechsc</a>

  </pre>
    </blockquote>
    <br>
  </blockquote>
</blockquote>
<br>
</body>
</html>

--------------080602050605060609010403--

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From: "Neil Deason" <ndeason@ubiquity.net>
To: "'Sarvi Shanmugham'" <sarvi@cisco.com>, <brian.wyld@eloquant.com>
Cc: "'Eric Burger'" <eburger@snowshore.com>,
        "'Brian Eberman'" <bse@speechworks.com>,
        "'IETF SPEECHSC \(E-mail\)'" <speechsc@ietf.org>
Subject: RE: [Speechsc] speechsc as a first-level protocol
Date: Fri, 9 May 2003 11:40:29 -0700
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=20

-----Original Message-----
From: speechsc-admin@ietf.org [mailto:speechsc-admin@ietf.org] On Behalf
Of Sarvi Shanmugham
Sent: 09 May 2003 09:49
To: brian.wyld@eloquant.com
Cc: 'Eric Burger'; 'Brian Eberman'; 'IETF SPEECHSC (E-mail)'
Subject: Re: [Speechsc] speechsc as a first-level protocol


Hi Brian,
       Great. I am half way through a proposal trying to do exactly
that. Since there was not much activity, I thought a proposal document
would drive some discussion. So after consulting with Eric, I have been
writing a proposal, doing exactly that.

As part 1, I have taken out all RTSP tunnelling and related aspects from
the currentt MRCP specification, this is done.  As Part to 2, I am
adding/advocating SIP based INVITE/BYE extensions a choice to create and
manage the session=20
=20
I am glad to see this part being included. I think this is the proper
relationship between speechsc and SIP.
=20
FYI I wrote a somewhat rough and ready draft on SDP usage for SOAP
sessions:
=20
=20
http://www.ietf.org/internet-drafts/draft-deason-mmusic-sdp-soap-00.txt
=20
If I can be of any help contributing to work on SIP/SDP for 'speechsc'
sessions please ask.
=20
Cheers,
Neil.


------=_NextPart_000_0007_01C3161F.CB7E38B0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1170" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  speechsc-admin@ietf.org [mailto:speechsc-admin@ietf.org] <B>On Behalf =
Of=20
  </B>Sarvi Shanmugham<BR><B>Sent:</B> 09 May 2003 09:49<BR><B>To:</B>=20
  brian.wyld@eloquant.com<BR><B>Cc:</B> 'Eric Burger'; 'Brian Eberman'; =
'IETF=20
  SPEECHSC (E-mail)'<BR><B>Subject:</B> Re: [Speechsc] speechsc as a =
first-level=20
  protocol<BR><BR></FONT></DIV>
  <DIV>Hi Brian,<BR>&nbsp; &nbsp; &nbsp; &nbsp;Great. I am half way =
through a=20
  proposal trying to do exactly that. Since there was not much activity, =
I=20
  thought a proposal document would drive some discussion. So after =
consulting=20
  with Eric, I have been writing a proposal, doing exactly =
that.<BR><BR>As part=20
  1, I have taken out all RTSP tunnelling and related aspects from the =
currentt=20
  MRCP specification, this is done.&nbsp; As Part to 2, I am =
adding/advocating=20
  SIP based INVITE/BYE extensions a choice to create and manage the =
session<SPAN=20
  class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2>&nbsp;</FONT></SPAN></DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New" =
size=3D2>I am glad=20
  to see this part being included. I think this is the proper =
relationship=20
  between speechsc and SIP.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D916593418-09052003></SPAN><SPAN=20
  class=3D916593418-09052003><FONT face=3D"Courier New" size=3D2>FYI I =
wrote a=20
  somewhat rough and ready draft on SDP usage for SOAP=20
  sessions:</FONT></SPAN></DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New" =
size=3D2>&nbsp; <A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-deason-mmusic-sdp-soap-=
00.txt">http://www.ietf.org/internet-drafts/draft-deason-mmusic-sdp-soap-=
00.txt</A></FONT></SPAN></DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New" =
size=3D2>If I can=20
  be of any help contributing to work on SIP/SDP for 'speechsc' sessions =
please=20
  ask.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2>Cheers,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D916593418-09052003><FONT face=3D"Courier New"=20
  size=3D2>Neil.</FONT></SPAN></DIV></BLOCKQUOTE></BODY></HTML>

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From: "Salvador Neto" <salvadon@exchange.microsoft.com>
To: "IETF SPEECHSC \(E-mail\)" <speechsc@ietf.org>,
        "Eric Burger" <eburger@snowshore.com>
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Subject: [Speechsc] SpeechSC and audio transmission protocols
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Hi all,

=20

We would like to raise the following concern on audio transmission
protocols in SpeechSC, which we had a chance to discuss with Eric Burger
a couple of weeks ago in person.

=20

We believe that SpeechSC as a control protocol should not define nor
dictate what media transmission protocol to use. People should be able
to use SpeechSC to negotiate different signal protocols (RTP or TCP for
example) according to different needs. In real time communication like
text-to-speech, RTP will be an appropriate choice. In a speech
recognition service scenario, a guaranteed-delivery protocol like TCP
could be more appropriate. One perspective on this is that there is a
trade-off between delay and quality of transmission that is dependent on
the application, and that SpeechSC itself should not decide that
trade-off, if it is to be applicable to different applications (and the
one that seems often overlooked in the discussions is a multimodal
client/DCR scenario).=20

=20

So, in summary, we believe the current requirement for SpeechSC to only
allow RTP is too heavy, and request that it be revisited in light of the
above.=20

=20

=20

Thanks,
Salva


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

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>We would&nbsp;like to raise the following concern on =
audio
transmission protocols in SpeechSC, which we had a chance to discuss =
with Eric
Burger a couple of weeks ago in person.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>We believe that SpeechSC as a control protocol should =
not
define&nbsp;nor dictate what&nbsp;media transmission&nbsp;protocol to
use.&nbsp;People should be able to use SpeechSC to negotiate different =
signal
protocols (RTP or TCP for example) according to different needs. In real =
time
communication like text-to-speech, RTP will be an appropriate choice. In =
a
speech recognition service scenario, a guaranteed-delivery protocol like =
TCP could
be&nbsp;more appropriate.&nbsp;One perspective on this is that there is =
a
trade-off&nbsp;between delay&nbsp;and quality of transmission that is =
dependent
on the application, and that SpeechSC itself should not decide that =
trade-off,
if it is to be applicable to different applications (and the one that =
seems
often overlooked in the discussions is a multimodal client/DCR =
scenario). </span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>So, in summary, we believe the current requirement
for&nbsp;SpeechSC to only allow RTP is&nbsp;too heavy,&nbsp;and request =
that it
be revisited in light of the above. </span></font></p>

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

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

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

</div>

</body>

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From: "Neil Deason" <ndeason@ubiquity.net>
To: "'Sarvi Shanmugham'" <sarvi@cisco.com>, <speechsc@ietf.org>
Subject: RE: [Speechsc] Protpcol proposal for SPEECHSC
Date: Wed, 28 May 2003 13:55:57 -0700
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Hi.

I have some comments on the usage of SDP for SPEECHSC part of the draft.


+ I would suggest an alternative format for the 'm=' line. Use a value
for the protocol that designates the underlying transport protocol, this
then allows for TCP, TLS, SCTP to be used with speechsc. For the media
format use the mime type 'application/speechsc'. Then use two SDP
attributes to describe the resource ID and channel ID. An example
written this way would therefore be:

		m=control 32416 TCP application/speechsc
		a=resourceId:speechrec
		a=channelId:32AECB234338 

+ Section 3.2 refers to setting the offerer setting the port to 0 in the
SDP when it wants to act as the client in the connection setup
procedure. However, setting a port to 0 has a defined meaning that a
stream is offered but must not be used according to RFC3264. This an
area where alignment with draft-ietf-mmusic-sdp-comedia-05.txt could
help. That recommends the offerer use a value of port 9 (the discard
port) in its offer. This work also then defines a SDP attribute to
define the client/server roles to be used in the connection set up
procedure, which could be reused here.

+ Section 3.2 refers to a client removing a 'm=' line from the SDP
description in a session to de-allocate a resource. RFC3264 describes
the mode for removing a media stream as continuing to offer it, but with
the port set to 0.

+ The SDP in Example 1 of the draft would not be legal according to RFC
2327 - the SDP in the INVITE is missing a mandatory 'm=' line. This may
not be a big deal as it was not clear to me why you would have this
preliminary offer/answer prior to establishing the resource control
channel. If there is no clear reason the exchange described in Example 1
would be unnecessary and can be removed.

+ Finally a few nits have crept in: The SIP URL used in the contact
header of the examples has an illegal whitespace in it, the SIP messages
are missing Via headers.

I have revised the examples in the draft accordingly below. They could
still use some cleaning up, but hopefully it helps illustrate the above
comments.

(Example 1 deleted - so pervious Example 2 becomes 1 etc.)

Example 1.
   C->S: 
           INVITE sip:mresources@mediaserver.com SIP/2.0 
           Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9 
           Max-Forwards: 70 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314161 INVITE 
           Contact: <sip:sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: ...
            
           v=0 
           o=sarvi 2890844526 2890842808 IN IP4 126.16.64.4 
           s=- 
           c=IN IP4 224.2.17.12
           m=control 9 TCP application/speechsc
	     a=direction:active 
           a=resourceId:speechsynth
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=recvonly 
     
    S->C: 
           SIP/2.0 200 OK 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314161 INVITE 
           Contact: <sip:sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: ... 
            
           v=0 
           o=sarvi 2890844526 2890842808 IN IP4 126.16.64.4 
           s=- 
           c=IN IP4 224.2.17.12
           m=control 32416 TCP application/speechsc
	     a=direction:passive
	     a=resourceId:speechsynth
	     a=channelId:32AECB234338 
           m=audio 48260 RTP/AVP 00 96 
           a=rtpmap:0 pcmu/8000 
           a=sendonly 
     
    C->S: 
           ACK sip:speechsc@mediaserver.com SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <sip:speechsc@mediaserver.com>;tag=a6c85cf 
           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314162 ACK 
           Content-Length: 0 

Example 2.
   C->S: 
           INVITE sip:mresources@mediaserver.com SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip:sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: ... 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 9 TCP application/speechsc
	     a=direction:active 
	     a=resourceId:speechrec
           m=control 9 TCP application/speechsc
	     a=direction:active 
	     a=resourceId:speechsynth
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=sendrecv 
     
    S->C: 
           SIP/2.0 200 OK 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip:sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: 131 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 32416 TCP application/speechsc
	     a=direction:passive
	     a=resourceId:speechrec
	     a=channelId:32AECB234338
           m=control 32416 TCP application/speechsc
	     a=channelId:32AECB234339
	     a=resourceId:speechsynth
	     a=direction:passive
           m=audio 48260 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=sendrecv 
     
    C->S: 
           ACK sip:speechsc@mediaserver.com SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <sip:speechsc@mediaserver.com>;tag=a6c85cf 
           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314164 ACK 
           Content-Length: 0 

Example 3.
    C->S: 
           INVITE sip:mresources@mediaserver.com SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip:sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: ...
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 0 TCP application/speechsc
	     a=direction:active
	     a=resourceId:speechrec
           m=control 9 TCP application/speechsc
	     a=direction:active
	     a=resourceId:speechsynth	
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=recvonly 
     
    S->C: 
           SIP/2.0 200 OK 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           To: MediaServer <sip:mresources@mediaserver.com> 
           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <sip:sarvi@cisco.com> 
           Content-Type: application/sdp 
           Content-Length: 131 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 0 TCP application/speechsc
	     a=direction:passive
	     a=resourceId:speechrec
	     a=channelId:32AECB234338
           m=control 32416 TCP application/speechsc
	     a=resourceId:speechsynth
	     a=channelId:32AECB234339
	     a=direction:passive
           m=audio 48260 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=sendonly 
     
    C->S: 
           ACK sip:speechsc@mediaserver.com SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <sip:speechsc@mediaserver.com>;tag=a6c85cf 
           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314164 ACK 
           Content-Length: 0 

Finally one thing I think the WG needs to be aware of is the proposed
new charter of the mmusic WG. A clear motivation there appears to be to
draw a line under what gets added to SDP as it is today and what would
be waiting for SDP NG. I think a too draconian interpretation of that
proposed charter might present a problem here. (As well as in the SIMPLE
WG).

hth,
Neil.
--
http://users.rcn.com/neildeason/

> -----Original Message-----
> From: speechsc-admin@ietf.org
> [mailto:speechsc-admin@ietf.org] On Behalf Of Sarvi Shanmugham
> Sent: 21 May 2003 14:30
> To: speechsc@ietf.org
> Subject: [Speechsc] Protpcol proposal for SPEECHSC
> 
> 
> 
> 
> Hi
>      Please find attached a proposal for a SPEECHSC protocol. This
> leverages the MRCP specification. But removes the tunnelling 
> aspects of 
> the protocol. I have also submitted this document to the 
> internet drafts 
> repository and should be available from there soon.
> 
> The current version does not address SI and SV aspects of the
> requirements document. But addresses most of the others. This 
> document 
> is meant to be a starting point for a possible direction that the 
> SPEECHSC protocol could take. Based on interest and comments 
> received, 
> we can proceed to add SI and SV capabilities in a following 
> version of 
> the draft.
> 
> Any thoughts.
> 
> Sarvi
> 

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



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Date: Thu, 29 May 2003 13:19:09 -0700
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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Subject: Re: [Speechsc] Protpcol proposal for SPEECHSC
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some comments line.

Neil Deason wrote:

>Hi.
>
>I have some comments on the usage of SDP for SPEECHSC part of the draft.
>
>
>+ I would suggest an alternative format for the 'm=' line. Use a value
>for the protocol that designates the underlying transport protocol, this
>then allows for TCP, TLS, SCTP to be used with speechsc. For the media
>format use the mime type 'application/speechsc'. Then use two SDP
>attributes to describe the resource ID and channel ID. An example
>written this way would therefore be:
>
>		m=control 32416 TCP application/speechsc
>		a=resourceId:speechrec
>		a=channelId:32AECB234338 
>
1.  I am fine with using TCP/SCTP/TLS as the transport field of the 
m-line and specifiying the channel-id/resource-id as an attribute of the 
m-line. That said, could you explain what we gain between using a 
transport of SPEECHSC(whisch specifies what transports are supported)  
and your suggestion of specifying TCP or SCTP in the transport field of 
the m-line. My current approach seems similar to examples used where the 
transport is claimed to be "h323".

2. Channel -id / resource format
Do you see problem with the channel id covering the resource identity 
and the channel identity?
I had done it this way, so that we have the option of requesting 
resource-id "00" to mean, on demand allocation. This returns a 
channel-id of  "32AECB23433800" where 00 means on-demand resource usage. 
There is no specific resourece that is being reserved or allocated.

This then allows an ASR or TTS or SI or SV command to be issued. The 
just need to have a channel-id of "32AECB234338XX" where XX is the 
resource identifier.

It also has the added benefit of figuring out from the chhanel -id what 
type of resource is being controlled.

>+ Section 3.2 refers to setting the offerer setting the port to 0 in the
>SDP when it wants to act as the client in the connection setup
>procedure. However, setting a port to 0 has a defined meaning that a
>stream is offered but must not be used according to RFC3264. This an
>area where alignment with draft-ietf-mmusic-sdp-comedia-05.txt could
>help. That recommends the offerer use a value of port 9 (the discard
>port) in its offer. This work also then defines a SDP attribute to
>define the client/server roles to be used in the connection set up
>procedure, which could be reused here.
>
Agreed. I will make the appropriate changes in the next revision.

>
>+ Section 3.2 refers to a client removing a 'm=' line from the SDP
>description in a session to de-allocate a resource. RFC3264 describes
>the mode for removing a media stream as continuing to offer it, but with
>the port set to 0.
>
Agreed. I will make the appropriate changes in the next revision.

>
>+ The SDP in Example 1 of the draft would not be legal according to RFC
>2327 - the SDP in the INVITE is missing a mandatory 'm=' line. This may
>not be a big deal as it was not clear to me why you would have this
>preliminary offer/answer prior to establishing the resource control
>channel. If there is no clear reason the exchange described in Example 1
>would be unnecessary and can be removed.
>
I was just trying to demonstrate that there could be times in the SIP 
session when there is no m-lines becuase there is no resources used or 
media between the client and the server. This may most likely happen mid 
session than at the beginning. the SIP session may add m-line for the 
different resources as need and m-lines for audio as needed by the 
resource. The client could allocate and deallocate resources(m-lines) as 
needed within in the session. This might mean that there are times, 
 when the client has released the all the individual resources but may 
want to allocate it as part of the session at a later point. At these 
points I don't want to have to destroy the whole SIP session and 
recreate it.
Anyway, atleast thats what I was trying to do. If this is illegal in 
SIP, I will remove it or find an alternative for this.

>
>+ Finally a few nits have crept in: The SIP URL used in the contact
>header of the examples has an illegal whitespace in it, the SIP messages
>are missing Via headers.
>
I will incorporate these into the next revision.

Thx,
Sarvi

>
>I have revised the examples in the draft accordingly below. They could
>still use some cleaning up, but hopefully it helps illustrate the above
>comments.
>
>(Example 1 deleted - so pervious Example 2 becomes 1 etc.)
>
>Example 1.
>   C->S: 
>           INVITE sip:mresources@mediaserver.com SIP/2.0 
>           Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9 
>           Max-Forwards: 70 
>           To: MediaServer <sip:mresources@mediaserver.com> 
>           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314161 INVITE 
>           Contact: <sip:sarvi@cisco.com> 
>           Content-Type: application/sdp 
>           Content-Length: ...
>            
>           v=0 
>           o=sarvi 2890844526 2890842808 IN IP4 126.16.64.4 
>           s=- 
>           c=IN IP4 224.2.17.12
>           m=control 9 TCP application/speechsc
>	     a=direction:active 
>           a=resourceId:speechsynth
>           m=audio 49170 RTP/AVP 0 96 
>           a=rtpmap:0 pcmu/8000 
>           a=recvonly 
>     
>    S->C: 
>           SIP/2.0 200 OK 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           To: MediaServer <sip:mresources@mediaserver.com> 
>           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314161 INVITE 
>           Contact: <sip:sarvi@cisco.com> 
>           Content-Type: application/sdp 
>           Content-Length: ... 
>            
>           v=0 
>           o=sarvi 2890844526 2890842808 IN IP4 126.16.64.4 
>           s=- 
>           c=IN IP4 224.2.17.12
>           m=control 32416 TCP application/speechsc
>	     a=direction:passive
>	     a=resourceId:speechsynth
>	     a=channelId:32AECB234338 
>           m=audio 48260 RTP/AVP 00 96 
>           a=rtpmap:0 pcmu/8000 
>           a=sendonly 
>     
>    C->S: 
>           ACK sip:speechsc@mediaserver.com SIP/2.0 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           Max-Forwards: 70 
>           To: MediaServer <sip:speechsc@mediaserver.com>;tag=a6c85cf 
>           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314162 ACK 
>           Content-Length: 0 
>
>Example 2.
>   C->S: 
>           INVITE sip:mresources@mediaserver.com SIP/2.0 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           Max-Forwards: 70 
>           To: MediaServer <sip:mresources@mediaserver.com> 
>           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314163 INVITE 
>           Contact: <sip:sarvi@cisco.com> 
>           Content-Type: application/sdp 
>           Content-Length: ... 
>            
>           v=0 
>           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
>           s=-
>           c=IN IP4 224.2.17.12
>           m=control 9 TCP application/speechsc
>	     a=direction:active 
>	     a=resourceId:speechrec
>           m=control 9 TCP application/speechsc
>	     a=direction:active 
>	     a=resourceId:speechsynth
>           m=audio 49170 RTP/AVP 0 96 
>           a=rtpmap:0 pcmu/8000 
>           a=rtpmap:96 telephone-event/8000 
>           a=fmtp:96 0-15 
>           a=sendrecv 
>     
>    S->C: 
>           SIP/2.0 200 OK 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           To: MediaServer <sip:mresources@mediaserver.com> 
>           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314163 INVITE 
>           Contact: <sip:sarvi@cisco.com> 
>           Content-Type: application/sdp 
>           Content-Length: 131 
>            
>           v=0 
>           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
>           s=-
>           c=IN IP4 224.2.17.12
>           m=control 32416 TCP application/speechsc
>	     a=direction:passive
>	     a=resourceId:speechrec
>	     a=channelId:32AECB234338
>           m=control 32416 TCP application/speechsc
>	     a=channelId:32AECB234339
>	     a=resourceId:speechsynth
>	     a=direction:passive
>           m=audio 48260 RTP/AVP 0 96 
>           a=rtpmap:0 pcmu/8000 
>           a=rtpmap:96 telephone-event/8000 
>           a=fmtp:96 0-15 
>           a=sendrecv 
>     
>    C->S: 
>           ACK sip:speechsc@mediaserver.com SIP/2.0 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           Max-Forwards: 70 
>           To: MediaServer <sip:speechsc@mediaserver.com>;tag=a6c85cf 
>           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314164 ACK 
>           Content-Length: 0 
>
>Example 3.
>    C->S: 
>           INVITE sip:mresources@mediaserver.com SIP/2.0 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           Max-Forwards: 70 
>           To: MediaServer <sip:mresources@mediaserver.com> 
>           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314163 INVITE 
>           Contact: <sip:sarvi@cisco.com> 
>           Content-Type: application/sdp 
>           Content-Length: ...
>            
>           v=0 
>           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
>           s=-
>           c=IN IP4 224.2.17.12
>           m=control 0 TCP application/speechsc
>	     a=direction:active
>	     a=resourceId:speechrec
>           m=control 9 TCP application/speechsc
>	     a=direction:active
>	     a=resourceId:speechsynth	
>           m=audio 49170 RTP/AVP 0 96 
>           a=rtpmap:0 pcmu/8000 
>           a=recvonly 
>     
>    S->C: 
>           SIP/2.0 200 OK 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           To: MediaServer <sip:mresources@mediaserver.com> 
>           From: sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314163 INVITE 
>           Contact: <sip:sarvi@cisco.com> 
>           Content-Type: application/sdp 
>           Content-Length: 131 
>            
>           v=0 
>           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
>           s=-
>           c=IN IP4 224.2.17.12
>           m=control 0 TCP application/speechsc
>	     a=direction:passive
>	     a=resourceId:speechrec
>	     a=channelId:32AECB234338
>           m=control 32416 TCP application/speechsc
>	     a=resourceId:speechsynth
>	     a=channelId:32AECB234339
>	     a=direction:passive
>           m=audio 48260 RTP/AVP 0 96 
>           a=rtpmap:0 pcmu/8000 
>           a=sendonly 
>     
>    C->S: 
>           ACK sip:speechsc@mediaserver.com SIP/2.0 
>	     Via: SIP/2.0/TCP
>client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>           Max-Forwards: 70 
>           To: MediaServer <sip:speechsc@mediaserver.com>;tag=a6c85cf 
>           From: Sarvi <sip:sarvi@cisco.com>;tag=1928301774 
>           Call-ID: a84b4c76e66710 
>           CSeq: 314164 ACK 
>           Content-Length: 0 
>
>Finally one thing I think the WG needs to be aware of is the proposed
>new charter of the mmusic WG. A clear motivation there appears to be to
>draw a line under what gets added to SDP as it is today and what would
>be waiting for SDP NG. I think a too draconian interpretation of that
>proposed charter might present a problem here. (As well as in the SIMPLE
>WG).
>
>hth,
>Neil.
>--
>http://users.rcn.com/neildeason/
>
>  
>
>>-----Original Message-----
>>From: speechsc-admin@ietf.org
>>[mailto:speechsc-admin@ietf.org] On Behalf Of Sarvi Shanmugham
>>Sent: 21 May 2003 14:30
>>To: speechsc@ietf.org
>>Subject: [Speechsc] Protpcol proposal for SPEECHSC
>>
>>
>>
>>
>>Hi
>>     Please find attached a proposal for a SPEECHSC protocol. This
>>leverages the MRCP specification. But removes the tunnelling 
>>aspects of 
>>the protocol. I have also submitted this document to the 
>>internet drafts 
>>repository and should be available from there soon.
>>
>>The current version does not address SI and SV aspects of the
>>requirements document. But addresses most of the others. This 
>>document 
>>is meant to be a starting point for a possible direction that the 
>>SPEECHSC protocol could take. Based on interest and comments 
>>received, 
>>we can proceed to add SI and SV capabilities in a following 
>>version of 
>>the draft.
>>
>>Any thoughts.
>>
>>Sarvi
>>
>>    
>>
>
>_______________________________________________
>Speechsc mailing list
>Speechsc@ietf.org
>https://www1.ietf.org/mailman/listinfo/speechsc
>
>  
>


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some comments line.<br>
<br>
Neil Deason wrote:<br>
<blockquote type="cite"
 cite="mid00d401c3255b$8ade9340$600119ac@eu.ubiquity.net">
  <pre wrap="">Hi.

I have some comments on the usage of SDP for SPEECHSC part of the draft.


+ I would suggest an alternative format for the 'm=' line. Use a value
for the protocol that designates the underlying transport protocol, this
then allows for TCP, TLS, SCTP to be used with speechsc. For the media
format use the mime type 'application/speechsc'. Then use two SDP
attributes to describe the resource ID and channel ID. An example
written this way would therefore be:

		m=control 32416 TCP application/speechsc
		a=resourceId:speechrec
		a=channelId:32AECB234338 </pre>
</blockquote>
1. &nbsp;I am fine with using TCP/SCTP/TLS as the transport field of the m-line
and specifiying the channel-id/resource-id as an attribute of the m-line.
That said, could you explain what we gain between using a transport of SPEECHSC(whisch
specifies what transports are supported)&nbsp; and your suggestion of specifying
TCP or SCTP in the transport field of the m-line. My current approach seems
similar to examples used where the transport is claimed to be "h323".<br>
    
<pre wrap="">
</pre>
2. Channel -id / resource format<br>
Do you see problem with the channel id covering the resource identity and
the channel identity?<br>
I had done it this way, so that we have the option of requesting resource-id
"00" to mean, on demand allocation. This returns a channel-id of &nbsp;"32AECB23433800"
where 00 means on-demand resource usage. There is no specific resourece that
is being reserved or allocated. <br>
<br>
This then allows an ASR or TTS or SI or SV command to be issued. The just
need to have a channel-id of "32AECB234338XX" where XX is the resource identifier.
<br>
<br>
It also has the added benefit of figuring out from the chhanel -id what type
of resource is being controlled.<br>
<blockquote type="cite"
 cite="mid00d401c3255b$8ade9340$600119ac@eu.ubiquity.net">
  <pre wrap="">
+ Section 3.2 refers to setting the offerer setting the port to 0 in the
SDP when it wants to act as the client in the connection setup
procedure. However, setting a port to 0 has a defined meaning that a
stream is offered but must not be used according to RFC3264. This an
area where alignment with draft-ietf-mmusic-sdp-comedia-05.txt could
help. That recommends the offerer use a value of port 9 (the discard
port) in its offer. This work also then defines a SDP attribute to
define the client/server roles to be used in the connection set up
procedure, which could be reused here.</pre>
</blockquote>
Agreed. I will make the appropriate changes in the next revision.<br>
<blockquote type="cite"
 cite="mid00d401c3255b$8ade9340$600119ac@eu.ubiquity.net">
  <pre wrap="">

+ Section 3.2 refers to a client removing a 'm=' line from the SDP
description in a session to de-allocate a resource. RFC3264 describes
the mode for removing a media stream as continuing to offer it, but with
the port set to 0.</pre>
</blockquote>
Agreed. I will make the appropriate changes in the next revision.<br>
<blockquote type="cite"
 cite="mid00d401c3255b$8ade9340$600119ac@eu.ubiquity.net">
  <pre wrap="">

+ The SDP in Example 1 of the draft would not be legal according to RFC
2327 - the SDP in the INVITE is missing a mandatory 'm=' line. This may
not be a big deal as it was not clear to me why you would have this
preliminary offer/answer prior to establishing the resource control
channel. If there is no clear reason the exchange described in Example 1
would be unnecessary and can be removed.</pre>
</blockquote>
I was just trying to demonstrate that there could be times in the SIP session
when there is no m-lines becuase there is no resources used or media between
the client and the server. This may most likely happen mid session than at
the beginning. the SIP session may add m-line for the different resources
as need and m-lines for audio as needed by the resource. The client could
allocate and deallocate resources(m-lines) as needed within in the session.
This might mean that there are times, &nbsp;when the client has released the all
the individual resources but may want to allocate it as part of the session
at a later point. At these points I don't want to have to destroy the whole
SIP session and recreate it.<br>
Anyway, atleast thats what I was trying to do. If this is illegal in SIP,
I will remove it or find an alternative for this.<br>
<blockquote type="cite"
 cite="mid00d401c3255b$8ade9340$600119ac@eu.ubiquity.net">
  <pre wrap="">

+ Finally a few nits have crept in: The SIP URL used in the contact
header of the examples has an illegal whitespace in it, the SIP messages
are missing Via headers.</pre>
</blockquote>
I will incorporate these into the next revision.<br>
<br>
Thx,<br>
Sarvi<br>
<blockquote type="cite"
 cite="mid00d401c3255b$8ade9340$600119ac@eu.ubiquity.net">
  <pre wrap="">

I have revised the examples in the draft accordingly below. They could
still use some cleaning up, but hopefully it helps illustrate the above
comments.

(Example 1 deleted - so pervious Example 2 becomes 1 etc.)

Example 1.
   C-&gt;S: 
           INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:mresources@mediaserver.com">sip:mresources@mediaserver.com</a> SIP/2.0 
           Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9 
           Max-Forwards: 70 
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:mresources@mediaserver.com">&lt;sip:mresources@mediaserver.com&gt;</a> 
           From: sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314161 INVITE 
           Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a> 
           Content-Type: application/sdp 
           Content-Length: ...
            
           v=0 
           o=sarvi 2890844526 2890842808 IN IP4 126.16.64.4 
           s=- 
           c=IN IP4 224.2.17.12
           m=control 9 TCP application/speechsc
	     a=direction:active 
           a=resourceId:speechsynth
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=recvonly 
     
    S-&gt;C: 
           SIP/2.0 200 OK 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:mresources@mediaserver.com">&lt;sip:mresources@mediaserver.com&gt;</a> 
           From: sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314161 INVITE 
           Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a> 
           Content-Type: application/sdp 
           Content-Length: ... 
            
           v=0 
           o=sarvi 2890844526 2890842808 IN IP4 126.16.64.4 
           s=- 
           c=IN IP4 224.2.17.12
           m=control 32416 TCP application/speechsc
	     a=direction:passive
	     a=resourceId:speechsynth
	     a=channelId:32AECB234338 
           m=audio 48260 RTP/AVP 00 96 
           a=rtpmap:0 pcmu/8000 
           a=sendonly 
     
    C-&gt;S: 
           ACK <a class="moz-txt-link-abbreviated" href="mailto:sip:speechsc@mediaserver.com">sip:speechsc@mediaserver.com</a> SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:speechsc@mediaserver.com">&lt;sip:speechsc@mediaserver.com&gt;</a>;tag=a6c85cf 
           From: Sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314162 ACK 
           Content-Length: 0 

Example 2.
   C-&gt;S: 
           INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:mresources@mediaserver.com">sip:mresources@mediaserver.com</a> SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:mresources@mediaserver.com">&lt;sip:mresources@mediaserver.com&gt;</a> 
           From: sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a> 
           Content-Type: application/sdp 
           Content-Length: ... 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 9 TCP application/speechsc
	     a=direction:active 
	     a=resourceId:speechrec
           m=control 9 TCP application/speechsc
	     a=direction:active 
	     a=resourceId:speechsynth
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=sendrecv 
     
    S-&gt;C: 
           SIP/2.0 200 OK 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:mresources@mediaserver.com">&lt;sip:mresources@mediaserver.com&gt;</a> 
           From: sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a> 
           Content-Type: application/sdp 
           Content-Length: 131 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 32416 TCP application/speechsc
	     a=direction:passive
	     a=resourceId:speechrec
	     a=channelId:32AECB234338
           m=control 32416 TCP application/speechsc
	     a=channelId:32AECB234339
	     a=resourceId:speechsynth
	     a=direction:passive
           m=audio 48260 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=rtpmap:96 telephone-event/8000 
           a=fmtp:96 0-15 
           a=sendrecv 
     
    C-&gt;S: 
           ACK <a class="moz-txt-link-abbreviated" href="mailto:sip:speechsc@mediaserver.com">sip:speechsc@mediaserver.com</a> SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:speechsc@mediaserver.com">&lt;sip:speechsc@mediaserver.com&gt;</a>;tag=a6c85cf 
           From: Sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314164 ACK 
           Content-Length: 0 

Example 3.
    C-&gt;S: 
           INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:mresources@mediaserver.com">sip:mresources@mediaserver.com</a> SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:mresources@mediaserver.com">&lt;sip:mresources@mediaserver.com&gt;</a> 
           From: sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a> 
           Content-Type: application/sdp 
           Content-Length: ...
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 0 TCP application/speechsc
	     a=direction:active
	     a=resourceId:speechrec
           m=control 9 TCP application/speechsc
	     a=direction:active
	     a=resourceId:speechsynth	
           m=audio 49170 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=recvonly 
     
    S-&gt;C: 
           SIP/2.0 200 OK 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:mresources@mediaserver.com">&lt;sip:mresources@mediaserver.com&gt;</a> 
           From: sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314163 INVITE 
           Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a> 
           Content-Type: application/sdp 
           Content-Length: 131 
            
           v=0 
           o=sarvi 2890844526 2890842809 IN IP4 126.16.64.4 
           s=-
           c=IN IP4 224.2.17.12
           m=control 0 TCP application/speechsc
	     a=direction:passive
	     a=resourceId:speechrec
	     a=channelId:32AECB234338
           m=control 32416 TCP application/speechsc
	     a=resourceId:speechsynth
	     a=channelId:32AECB234339
	     a=direction:passive
           m=audio 48260 RTP/AVP 0 96 
           a=rtpmap:0 pcmu/8000 
           a=sendonly 
     
    C-&gt;S: 
           ACK <a class="moz-txt-link-abbreviated" href="mailto:sip:speechsc@mediaserver.com">sip:speechsc@mediaserver.com</a> SIP/2.0 
	     Via: SIP/2.0/TCP
client.atlanta.example.com:5060;branch=z9hG4bK74bf9
           Max-Forwards: 70 
           To: MediaServer <a class="moz-txt-link-rfc2396E" href="mailto:sip:speechsc@mediaserver.com">&lt;sip:speechsc@mediaserver.com&gt;</a>;tag=a6c85cf 
           From: Sarvi <a class="moz-txt-link-rfc2396E" href="mailto:sip:sarvi@cisco.com">&lt;sip:sarvi@cisco.com&gt;</a>;tag=1928301774 
           Call-ID: a84b4c76e66710 
           CSeq: 314164 ACK 
           Content-Length: 0 

Finally one thing I think the WG needs to be aware of is the proposed
new charter of the mmusic WG. A clear motivation there appears to be to
draw a line under what gets added to SDP as it is today and what would
be waiting for SDP NG. I think a too draconian interpretation of that
proposed charter might present a problem here. (As well as in the SIMPLE
WG).

hth,
Neil.
--
<a class="moz-txt-link-freetext" href="http://users.rcn.com/neildeason/">http://users.rcn.com/neildeason/</a>

  </pre>
  <blockquote type="cite">
    <pre wrap="">-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:speechsc-admin@ietf.org">speechsc-admin@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:speechsc-admin@ietf.org">mailto:speechsc-admin@ietf.org</a>] On Behalf Of Sarvi Shanmugham
Sent: 21 May 2003 14:30
To: <a class="moz-txt-link-abbreviated" href="mailto:speechsc@ietf.org">speechsc@ietf.org</a>
Subject: [Speechsc] Protpcol proposal for SPEECHSC




Hi
     Please find attached a proposal for a SPEECHSC protocol. This
leverages the MRCP specification. But removes the tunnelling 
aspects of 
the protocol. I have also submitted this document to the 
internet drafts 
repository and should be available from there soon.

The current version does not address SI and SV aspects of the
requirements document. But addresses most of the others. This 
document 
is meant to be a starting point for a possible direction that the 
SPEECHSC protocol could take. Based on interest and comments 
received, 
we can proceed to add SI and SV capabilities in a following 
version of 
the draft.

Any thoughts.

Sarvi

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