[SIPForum-discussion] DTMF

Wed May 23 06:29:38 UTC 2007

Hi Praveen,

Consider this scenario -

Step-1. A user calls a call center by dialing 1600-4444-1111
Step-2. The call lands on to an IVR system and prompts for further digital 
choices.
Step-3. The user further dials 1-1-1-9 through 4 levels of voice prompt

Step-1 above:
If the user uses an analog phone, i.e., a POTS phone connected to the 
local exchange via old-styled copper wire pair, then whatever the user 
keys in (before or after the IVR systems comes into picture) goes to the 
exchange in form of specific frequencies (tone or pulse). The exchange 
understands and initially sends an ISUP message (IAM) to the destination 
voip gateway (that connects to the SIP side/call center)

After the GW receives the IAM message, it may create a SIP INVITE message 
with any of the following ways
        - tel:160044441111 at domain.com
        - sip: 1600-4444-1111 at domain.com 
                (Above 2 cases: number gets further resolved, if there is 
app-server beyond the GW in the network)
        - GW uses ENUM and finds a IP address/FQDN for the destination and 
uses the same with sip:

Step-2 above:
As soon as the IVR system is reached, an end to end media path is setup. 
Hence the user hears prompts from the system

Step-3 above:
If the user dials digits following IVR prompt, to the local exchange the 
information goes, as it would have gone in Step-1. Exchange then carries 
this digit further in mid-call signaling procedure (FACILITY) to the GW. 
GW may understand that a Voice path exist. It has two ways to carry the 
DTMF infotmation forward to App-server/UA
        1. RTP payload with RFC2833 header 
(Content-type=application/telephony-event)
        2. Sends a SIP INFO method with DTMF info in the body 
(Content-type=application/dtmf)

RFC2833 should provide detailed info you may be interested in.

thanks,
Dipanjan

"praveen subbarao" <praveen_subbarao at rediffmail.com> 
Sent by: discussion-bounces at sipforum.org
05/23/2007 11:23 AM

Please respond to
praveen subbarao <praveen_subbarao at rediffmail.com>

To
"Eric Burger" <eburger at bea.com>
cc
discussion <discussion at sipforum.org>
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
> >
> >
> >
> > <
http://adworks.rediff.com/cgi-bin/AdWorks/click.cgi/www.rediff.com/signature-

> > 
home.htm/1050715198 at Middle5/1189676_1183578/1188797/1?PARTNER=3&OAS_QUERY=
> > null target=new>
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
> >
> > ***********************  Aricent-Private  ***********************
> > "DISCLAIMER: This message is proprietary to Aricent and is intended 
solely for
> > the use of
> > the individual to whom it is addressed. It may contain privileged or
> > confidential information and should not be
> > circulated or used for any purpose other than for what it is intended. 
If you
> > have received this message in error,
> > please notify the originator immediately. If you are not the intended
> > recipient, you are notified that you are strictly
> > prohibited from using, copying, altering, or disclosing the contents 
of this
> > message. Aricent accepts no responsibility for
> > loss or damage arising from the use of the information transmitted by 
this
> > email including damage from virus."
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
>
>
>
>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.

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit 
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org

***********************  Aricent-Private   ***********************
"DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of 
the individual to whom it is addressed. It may contain privileged or confidential information and should not be 
circulated or used for any purpose other than for what it is intended. If you have received this message in error, 
please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly
prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for 
loss or damage arising from the use of the information transmitted by this email including damage from virus."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20070523/7f747558/attachment-0002.html>