Internet Communications Using SIP : Delivering VoIP and Multimedia Services With Session Initiation Protocol {2Nd Ed.}

(Steven Felgate) #1

Collecting DTMF Digits


Figure 19.5 shows the basic call flows for the plain collecting of dual-tone
multi-frequency digits, without voice recognition. We will discuss how SIP
third-party call control is used for this application.
The initial INVITE(Message 1 in Figure 19.5) from the caller is directed to
the service controller. The Request-URI in the INVITEmessage identifies this
service, so various SIP proxy servers in the network (not shown here) know to
route the call to the controller.
The controller first forwards the INVITEto the DTMF collector (Message 2)
with no SDP body. This creates an initial media stream β€œon hold.” The DTMF
collector answers with its own SDP body in the reply 200 OK(Message 3). The
controller uses the reply (Message 3) to capture the data in the SDP body from
the DTMF collector. It then proxies the call to the desired called party in Mes-
sage 5 and gets, in return, a 200 OK(Message 6) in case of success. The
response to the caller (Message 7) has the following form:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.101.102.103;branch=z9hG4bK7d
To: User A <sip:[email protected]>;tag=3422
From: User B <sip:[email protected]>;tag=81211
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: ...

v=0
o=UserA 289375749 289375749 IN IP5 110.111.112.113
s=-
c=IN IP4 110.111.112.113
t=0 0
m=audio 5004 RTP/AVP 0

After Message 9 in Figure 19.5, the caller and called party can communicate.
Possible IP-PSTN VoIP gateways are not shown for clarity of the SIP call flows
on the IP side.

330 Chapter 19

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