[SIPForum-discussion] DTMF

Wed May 23 08:04:39 UTC 2007

Hi Srinivas,

Thanks for the info. The gist is:

* Enum is used for a direct calling from PSTN to SIP client
* RTP packets carry DTMF for mid-call dialling.

Thanks,

Praveen  

On Wed, 23 May 2007 Srinivas. wrote :
>
>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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