[SIPForum-discussion] DTMF

Wed May 23 09:33:05 UTC 2007

Richard,

Thanks for the clarification.

________________________________

From: Stastny Richard [mailto:Richard.Stastny at oefeg.at] 
Sent: Wednesday, May 23, 2007 1:50 PM
To: Srinivas .; praveen subbarao; Eric Burger
Cc: discussion
Subject: RE: [SIPForum-discussion] DTMF

To clarify:

>Case 1. PSTN user just picks the phone and dials the number, here the
dialed numbers will be passed in 

>Request URI / To fields of INVITE.

>Request and To headers will be in the form of tel uri. (It may be in
the form of E.164 number format, 
>depends on the number dialed).

Dialstrings should in future be passed in the Request URI in the format
according
to draft-rosen-iptel-dialstring-05.txt, e.g:

sip:123;phone-context=atlanta.example.com at homeprovider.com;user=dialstri
ng

The problem with the (local) format of the tel: URI is that it does not
allow * and #

the global format of the tel: URI may be used, but not for dialed
digits:
you cannot dial E.164 numbers.  tel:+12345678900

E.164 numbers may only be entered  by UA capable of entering a '+". 

So a normal PSTN phone with a terminal
adapter can never enter an E.164 number direct.

ENUM can only be used by the UA if it capable of receiving
an E.164 number (with a '+') or has a local dialstring processing
(and has the capability to query ENUM)

Regards
Richard

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Srinivas .
Sent: Wednesday, May 23, 2007 8:38 AM
To: praveen subbarao; Eric Burger
Cc: discussion
Subject: Re: [SIPForum-discussion] DTMF

To avoid confusion let me put it as two different cases.

Case 1. PSTN user just picks the phone and dials the number, here the
dialed numbers will be passed in Request URI / To fields of INVITE.

            Request and To headers will be in the form of tel uri. (It
may be in the form of E.164 number format, depends on the number
dialed).

Case 2. Suppose the call is established and user hear some announcements
(IVR) to get some more info. In such cases (mid call dialing) the dialed
numbers are carried in RTP (payload type as DTMF).

Am I correct?

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of praveen subbarao
Sent: Wednesday, May 23, 2007 11:23 AM
To: Eric Burger
Cc: discussion
Subject: Re: [SIPForum-discussion] DTMF

Hi Eric,

Thanks for the reply. What i understand is that, the digits will be
converted to some form of address & will be put in INVITE. Is it some
thing called ENUM?? Let me know

Praveen

On Tue, 22 May 2007 Eric Burger wrote :
>It looks like Praveen is asking how the dialed-number gets conveyed.
The
>whole point is digits do not traverse the network; addresses do.  See
SS7
>ISUP for how this works in the PSTN.  Either the phone (if SIP-enabled)
or
>gateway collects the digits and formulates the target address and puts
it
>into an INVITE.
>
>
>On 5/21/07 3:52 AM, "dipanjan.dutta at aricent.com"
><dipanjan.dutta at aricent.com> wrote:
>
> >
> > Hi Praveen,
> >
> > DTMF information gets exchange after the media patch is  connected.
It is also
> > called in-band signaling,
> > The media path can be connected either through early media
connection or
> > through 200-ACK.
> >
> > Further if any of the endpoint presses any digit in DTMF mode, this
> > information is conveyed over media path using RTP. RTP header will
indicate
> > DTMF payload and inside the payload, the DTMF event would be encoded
using RFC
> > 2833.
> >
> > While the UAs do initial signaling, they can a priori do a media
negotiation
> > for audio/telephone-event MIME type through SDP, so that anytime
afterwards
> > during a call, direct switching to DTMF in RTP can go on w/o issues.
> >
> > thanks,
> > Dipanjan
> >
> >
> >
> >
> > "praveen subbarao" <praveen_subbarao at rediffmail.com>
> > Sent by: discussion-bounces at sipforum.org 05/21/2007 11:56 AM
> > Please respond to
> > praveen subbarao <praveen_subbarao at rediffmail.com>
> > To
> > "discussion" <discussion at sipforum.org>
> > cc
> > Subject
> > [SIPForum-discussion] DTMF
> >
> >
> >
> >
> > Hi All,
> >
> > I have a doubt in SIP. Supposing that a SIP client wish to
communicate with a
> > PSTN client, then he will input the PSTN number.
> >
> > In SIP, where & how the DTMF information is carried. Which are the
headers
> > responsible for this? I rquest to knidly clarify.
> >
> > Regards,
> > Praveen
> >
> >
> >
> >
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