List Info

Thread: Re: INFO




Re: INFO
user name
2007-09-10 15:26:23
Precisely. To be blunt, one would be "less than
smart" to use either KPML or INFO for any interactive
voice response application.

INFO doesn't route by default. You have to *hope* your
softswitch, SBC, or other B2BUA in the way knows to relay
the INFO. At least most of those elements knows to relay
pub/sub.

Both KPML and INFO have the proxy latency penalty, making
for a piss-poor user experience.  

While we're at it, let's transport voice in SIP 

--
Sent from my wireless e-mail device. Sorry if terse.  We all
need lemonade: see <http:
//www.standardstrack.com/ietf/lemonade> for what
lemonade is.

----- Original Message -----
From: Francois Audet <audetnortel.com>
To: Hadriel Kaplan <HKaplanacmepacket.com>; Brian
Stucker <bstuckernortel.com>; Eric Burger
Cc: sip <sipietf.org>
Sent: Mon Sep 10 13:08:58 2007
Subject: RE: [Sip] INFO

Ah.... Good point. Hadn't tought of that.

In any case, I'm not sure voicemail is the best example.
Seems to me
you are more likely to use RFC 2833 for voicemail, since
it's a 
media server. Furthermore, you'll for sure be dealing with
people 
leaving voicemails that are NOT SIP users and would be going
through
a generic PSTN-SIP gateway (which would do RFC 2833).

I'd think that KPML is more useful for features that don't
terminate
media, like the Calling card example, or *gasp* invocation
of feature 
codes, and where the end-user device is known to be a SIP
phone.


> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplanacmepacket.com] 
> Sent: Monday, September 10, 2007 12:42
> To: Audet, Francois (SC100:3055); Stucker, Brian 
> (RICH1:AR00); 'Eric Burger'
> Cc: 'sip'
> Subject: RE: [Sip] INFO
> 
> But most calls to a voicemail server DON'T actually use
DTMF, 
> I've been told
> - since most calls to a vmail server are to leave
messages.  
> Calls to retrieve voicemail do always use dtmf, but
calls to 
> leave voicemail don't.
> However calls to leave voicemail usually have the
optional 
> ability for the caller to send DTMF - for example
pressing 
> "*" and then the passcode to retrieve
voicemail, or "0" to 
> reach an attendant, or some button to cancel the
recording, 
> etc.  Only a fraction of those calls actually end up
with 
> someone pressing a dtmf button, but because you *could*
press 
> dtmf, the vmail server would have to KPML subscribe for
all calls.

Notice:  This email message, together with any attachments,
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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sip

RE: INFO
country flaguser name
Canada
2007-09-10 15:38:53
 

> -----Original Message-----
> From: Eric Burger [mailto:eburgerbea.com] 
> Sent: Monday, September 10, 2007 3:26 PM
> To: Audet, Francois (SC100:3055); HKaplanacmepacket.com; 
> Stucker, Brian (RICH1:AR00)
> Cc: sipietf.org
> Subject: Re: [Sip] INFO
> 
> Precisely. To be blunt, one would be "less than
smart" to use 
> either KPML or INFO for any interactive voice response
application.
> 
> INFO doesn't route by default. You have to *hope* your

> softswitch, SBC, or other B2BUA in the way knows to
relay the 
> INFO. At least most of those elements knows to relay
pub/sub.
> 
> Both KPML and INFO have the proxy latency penalty,
making for 
> a piss-poor user experience.  

Can you idenitfy ANY IVR application that people didn't
think it was a
poor user experience? Yeah, IVR jail. Wheee...

> 
> While we're at it, let's transport voice in SIP 
> 
> --
> Sent from my wireless e-mail device. Sorry if terse. 
We all 
> need lemonade: see 
> <http:
//www.standardstrack.com/ietf/lemonade> for what
lemonade is.
> 
> ----- Original Message -----
> From: Francois Audet <audetnortel.com>
> To: Hadriel Kaplan <HKaplanacmepacket.com>; Brian
Stucker 
> <bstuckernortel.com>; Eric Burger
> Cc: sip <sipietf.org>
> Sent: Mon Sep 10 13:08:58 2007
> Subject: RE: [Sip] INFO
> 
> Ah.... Good point. Hadn't tought of that.
> 
> In any case, I'm not sure voicemail is the best
example. 
> Seems to me you are more likely to use RFC 2833 for 
> voicemail, since it's a media server. Furthermore,
you'll for 
> sure be dealing with people leaving voicemails that are
NOT 
> SIP users and would be going through a generic PSTN-SIP

> gateway (which would do RFC 2833).
> 
> I'd think that KPML is more useful for features that
don't 
> terminate media, like the Calling card example, or
*gasp* 
> invocation of feature codes, and where the end-user
device is 
> known to be a SIP phone.
> 
> 
> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplanacmepacket.com] 
> > Sent: Monday, September 10, 2007 12:42
> > To: Audet, Francois (SC100:3055); Stucker, Brian 
> > (RICH1:AR00); 'Eric Burger'
> > Cc: 'sip'
> > Subject: RE: [Sip] INFO
> > 
> > But most calls to a voicemail server DON'T
actually use DTMF, 
> > I've been told
> > - since most calls to a vmail server are to leave
messages.  
> > Calls to retrieve voicemail do always use dtmf,
but calls to 
> > leave voicemail don't.
> > However calls to leave voicemail usually have the
optional 
> > ability for the caller to send DTMF - for example
pressing 
> > "*" and then the passcode to retrieve
voicemail, or "0" to 
> > reach an attendant, or some button to cancel the
recording, 
> > etc.  Only a fraction of those calls actually end
up with 
> > someone pressing a dtmf button, but because you
*could* press 
> > dtmf, the vmail server would have to KPML
subscribe for all calls.
> 
> Notice:  This email message, together with any
attachments, 
> may contain information  of  BEA Systems,  Inc.,  its 
> subsidiaries  and  affiliated entities,  that may be 
> confidential,  proprietary,  copyrighted  and/or
legally 
> privileged, and is intended solely for the use of the 
> individual or entity named in this message. If you are
not 
> the intended recipient, and have received this message
in 
> error, please immediately return this by email and then
delete it.
> 


_______________________________________________
Sip mailing list  https://ww
w1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementorscs.columbia.edu for questions on current
sip
Use sippingietf.org for new developments on the application of
sip

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