1.3 Overview

  • Article

This protocol adds extensions to the Session Initiation Protocol (SIP), for interfacing a protocol client with other traditional telephony networks, such as a public switched telephone network (PSTN) and an enterprise private branch exchange (PBX) or IP-PBX.

The logical entities that are affected by these extensions are protocol client, server (proxy), and gateway entities. The protocol client and the gateway can function as a user agent client (UAC) or user agent server (UAS), depending on their role in the SIP transaction, as illustrated in the following diagram.

SIP transaction

Figure 1: SIP transaction

The extensions do the following:

  • Enable a SIP user agent (SIP UA) to be aware that a remote SIP UA in a SIP dialog is a gateway, as described in section 2.2.1 and section 3.1. This information can be rendered to the user interface (UI) to provide a better user experience (UX).

  • Enable a SIP URI to hold an address of a dial string that is given by a user, as described in section 2.2.2 and section 3.2.

  • Enable a SIP UAS to detect a redundant call that is triggered as a result of a loop, as described in section 2.2.3 and section 3.3. A loop occurs when a call is forked to a PBX that forks the call back, using a new SIP dialog.

  • Enable a SIP UA to indicate that it is willing to receive an SDP answer through a non-reliable 183 provisional response to an INVITE message, as described in section 2.2.4 and section 3.4. Note that the standard recommends sending an SDP answer for early media only through a reliable provisional response, as described in [RFC3262].

  • Define an anonymous phone URI, as described in section 2.2.5 and section 3.5, as an alternative to the standard anonymous SIP URI, as described in [RFC3261]. Note that the standard anonymous SIP URI is not supported.

  • Enable a SIP UA in the protocol network to indicate that it supports media bypass functionality, as described in section 2.2.6 and section 3.6. Media bypass has the media from the protocol network entity involved in a PSTN call going directly to the gateway used to interface with the PSTN for that call, without traversing any intermediate element in the protocol network.

  • Enable a SIP UA in the protocol network to reference the appropriate Session Description Protocol (SDP) that was selected from a received offer when sending a SIP message with an answer to the offer, as described in section 2.2.7 and section 3.7.

  • Identify the specific gateway used to interface with the PSTN for a PSTN call, as described in section 2.2.8 and section 3.8.