The IAM can be mapped to the SIPINVITEMessage 2 of Figure 11.3 as
shown here:INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.carrier.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: <sip:[email protected];user=phone>tag=3342k
To: <sip:[email protected];user=phone>
Call-ID: 123456028796867655
CSeq: 10 INVITE
Contact: <sip:gw1.carrier.com>
Content-Type: application/sdp
Content-Length: 156v=0
o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.carrier.com
s=-
c=IN IP4 gatewayone.carrier.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000Some field mapping is obvious, such as calling party number to From, but
others are not so obvious. In particular, there are generally many more para-
meters in a PSTN signaling message than can be mapped to a SIP message.
(For example, how is the forward call indicator mapped to SIP?) The result is
some information loss. However, if the call routes over the SIP network to the
destination, there is no net effect on the call completion, since all information
usable in the SIP network has been mapped. The additional parameters that
are not mapped from ISUP to SIP are designed for PSTN routing, not SIP rout-
ing, and their loss has no effect.
Similarly, mapping from SIP to ISUP (shown in Figure 11.2) does not cause
a loss in functionality. In this case, some ISUP parameters that have no coun-
terpart in SIP will need to be created for the mapped IAM. These values are set
to default values, typically on a trunk group basis.
However, should a call be routed from the PSTN to SIP, and then back to the
PSTN, some of the lost parameters from the first PSTN leg could be useful in
routing in the second PSTN leg. To solve this problem for networks designed
to do this, the encapsulation of PSTN signaling messages, in addition to inter-
networking, was developed. This application of SIP, known as SIP Telephony
(SIP-T) [1], carries the PSTN signaling information in the SIP signaling mes-
sage as a MIME message body [6]. The terminating gateway then constructs
the second leg PSTN signaling based on the SIP signaling parameters and the
attached message body of the original PSTN signaling. The resulting network
offers the possibility of making the SIP leg of the call transparent to the PSTN.
Put another way, SIP-T enables the ISUP transparency across a SIP network.
This call flow is shown in Figure 11.4.192 Chapter 11