to be encrypted using S/MIME because of the sensitivity of some information
present there (such as the calling party number of a private call), leading to
additional complexity in gateways. Alternatively, a network of screening prox-
ies will be needed to selectively remove ISUP attachments. (These proxies will
have to decode the multipart MIME attachment composed of the SDP and the
ISUP, extract only the SDP, recalculate the octet count, and then forward the
message.) This adds delay and complexity to the SIP network.
The simplest argument against SIP-T is the fact that SIP phones cannot eas-
ily talk to SIP-T gateways, and there are now really two types of SIP endpoints
that cannot talk to each other. As a result, T-SIP gateways can talk only directly
to other SIP-T gateways, and SIP phones can talk only to other SIP phones.
Some early SIP networks will implement SIP-T, especially in so-called
“softswitch” networks. However, for truly scalable and cost-effective tele-
phony, the complexity and protocols of the PSTN must not be carried into the
IP domain—a true SIP/PSTN internetworking gateway is required and ISUP
tunneling or encapsulation is not required.
A summary of SIP-to-PSTN protocol mapping is shown in Table 11.2. Note
that this table is greatly simplified and the actual mapping is more complex,
depending on the version of ISUP used (ETSI, ANSI, and so forth).
Table 11.2 SIP-to-ISUP and ISDN Message Mapping
SIP MESSAGE OR RESPONSE ISUP MESSAGE ISDN MESSAGE
INVITE IAM or SAM Setup
INFO USR User
BYE REL Release
CANCEL REL Release
ACK — —
REGISTER — —
18x ACM or CPG Alerting
200 (to INVITE) ANM or CON Connect
4xx, 5xx, 6xx REL Release
200 (to BYE) RLC Release CompleteSIP Telephony 195