FAQ: What are SIP-I and SIP-T?

SIP-I and SIP-T refer to two very similar approaches for interworking ISUP networks with SIP networks. In particular, they provide the means for conveying ISUP-specific parameters through a SIP network so that calls that originate and terminate on the ISUP network can transit a SIP network with no loss of information. SIP-T was developed by the IETF the same body that developed the SIP protocol itself around the same time the most recent version of SIP was being developed (mid-2002). It is defined by RFC 3372, RFC 3398, RFC 3578, and RFC 3204. SIP-I was developed by the ITU in 2004, and made use of most of the constructs defined in the IETF SIP-T effort. It is defined by ITU-T Q.1912.5. SIP-I and SIP-T both define the mapping of messages, parameters, and error codes between SIP and ISUP. Both of them are fully interoperable with compliant SIP network components on the SIP network. The key differences between SIP-I and SIP-T are: 1. SIP-I defines a mapping from SIP to BICC (in additional to ISUP), while SIP-T addresses only the ISUP case, and 2. SIP-T is inherently designed for interoperation with native SIP terminals, while SIP-I is restricted for use between PSTN gateways only. SIP-I and SIP-T also define somewhat different mappings of information between the protocols, mostly in terms of converting from SIP error codes to ISUP cause codes. The way SIP-I and SIP-T allow transparent transit of ISUP parameters through a SIP network is by attaching a literal copy of the original ISUP message to the SIP message at the ingress PSTN gateway; this ISUP message appears as another body on the SIP message (typically, a peer to an SDP body). The SIP network ignores the extra ISUP body, processing the SIP message as it normally would. After the SIP service network performs any necessary modifications to the SIP message, it arrives at the PSTN egress gateway. This egress gateway uses the attached ISUP message as the basis for the ISUP message it will send; however, it first makes modifications necessary to match changes made to the SIP message during its traversal of the SIP network.

As mentioned before, with SIP-T, the messages may also terminate on the native SIP terminals in the network, which will ignore the extra ISUP body. Additionally, messages may originate from these SIP phones and terminate on the PSTN gateways, which will then generate a new ISUP message for the PSTN.

Putting this together in a call flow, a typical successful call setup from a PSTN terminal to another PSTN terminal through a SIP network can look something like this: