Session Initiation Protocol with encapsulated ISDN User Part

Description

SIP-I is a critical interworking protocol defined by 3GPP and ITU-T (as Q.1912.5) that facilitates communication between Session Initiation Protocol (SIP) based Next Generation Networks (NGN) and the legacy Public Switched Telephone Network (PSTN) which uses the ISDN User Part (ISUP) signaling protocol. Its primary function is to ensure that call control signaling and supplementary services can be translated and transported between these disparate network domains without loss of functionality. The protocol achieves this by encapsulating entire ISUP messages, including their parameters and information elements, within the body of SIP messages, typically using the MIME type `application/isup`. This encapsulation preserves the full semantic content of the ISUP signaling, allowing for transparent end-to-end service delivery.

Architecturally, SIP-I operates at the signaling border between networks, such as at a Media Gateway Controller (MGC) or a Session Border Controller (SBC) that interfaces an IP Multimedia Subsystem (IMS) network with a legacy circuit-switched core. When a call originates from the PSTN destined for an IMS user, the originating network's signaling point sends an ISUP Initial Address Message (IAM). The interworking function receives this ISUP message, maps the critical call setup information (like called/calling party numbers) into corresponding SIP header fields (like To and From), and then embeds the original, binary-encoded ISUP message into the body of a SIP INVITE request. This SIP INVITE is then routed through the IP network. The receiving interworking function on the other side of the IP network extracts the encapsulated ISUP message and uses it to generate the appropriate ISUP signaling towards the terminating PSTN leg, ensuring feature transparency.

The protocol's design includes specific mapping rules between ISUP parameters and SIP headers, as well as procedures for handling subsequent call control messages like Answer (ANM), Release (REL), and various supplementary service invocations. It supports both "encapsulation" mode, where the full ISUP message is carried, and "translation" mode, where only the semantically equivalent SIP information is used, though encapsulation is preferred for maximum compatibility. SIP-I's role is foundational for network operators transitioning to all-IP cores like IMS, as it allows them to maintain existing PSTN/ISDN services and inter-carrier agreements while modernizing their infrastructure. It ensures that complex telephony features like caller ID, call forwarding, and closed user group services work seamlessly across the hybrid network environment.

Purpose & Motivation

SIP-I was created to solve the critical problem of interworking between the emerging, packet-based IP Multimedia Subsystem (IMS) and the vast, entrenched infrastructure of the legacy circuit-switched PSTN. As operators began deploying IMS in the mid-2000s to deliver voice and multimedia services over IP, they faced the immediate challenge of connecting these new IP networks to the existing global telephone network. Simple protocol translation was insufficient because ISUP carries rich, complex signaling information for call control and supplementary services that early, basic SIP-to-ISUP mapping could not fully preserve, leading to service degradation or failure for cross-network calls.

The historical context is the long, gradual transition from TDM-based networks to all-IP networks. Prior approaches, like simple SIP-to-ISUP protocol conversion, often resulted in the loss of non-basic call information elements, breaking advanced telephony services. This created a major barrier to IMS adoption. SIP-I was motivated by the need for a robust, standardized method that guaranteed feature transparency, ensuring that a call originating in the PSTN and terminating in IMS (or vice versa) would have access to the same set of services as a call that remained entirely within one network domain. Its creation was driven by both 3GPP and ITU-T to provide a unified, reliable specification that equipment vendors and network operators could implement to ensure global interoperability during the multi-decade migration path from legacy SS7 to IP-based signaling.

Key Features

• Full encapsulation of binary ISUP messages within SIP message bodies (MIME type application/isup)
• Support for transparent end-to-end carriage of ISUP signaling information for call control and supplementary services
• Defined mapping rules between ISUP parameters (e.g., calling party category) and SIP header fields and parameters
• Procedures for handling call establishment, answer, release, and error scenarios across the interworking boundary
• Compatibility with both 3GPP IMS networks and ITU-T NGN architectures
• Enables service transparency for features like number portability, caller ID, and call forwarding in hybrid networks

Evolution Across Releases

Defining Specifications