SS7/IP Interworking Tutorial
- SIP, PINT, SPIRITS, ENUM, TRIP

SIP (Session Initiation Protocol) is a signaling protocol for creating, modifying and terminating sessions, such as IP voice calls or multimedia conferences, with one or more participants in an IP network. SIP is currently undergoing standardization by the Internet Engineering Task Force SIP Working Group. While the sigtran protocols are currently the protocols of choice for interworking IP networks with the public switched telephone network, SIP is the protocol of choice for converged communications networks in the near future.

SIP provides the following functions:

• Name Translation and User Location: to ensure that a call reaches the called party regardless of location. SIP addresses users by an email-like address. Each user is identified through a hierarchical URL built around elements such as a user's telephone number or host name (for example, SIP:user@company.com). Because of this similarity, SIP URLs are easy to associate with a user's e-mail address.
  
  
• Feature Negotiation: SIP allows all parties involved in a call to negotiate and agree on the features supported, recognizing that all participants may not be able to support the same kind of features. For example, a session between a mobile voice-only telephone user and two video-enabled device users would agree to support voice features only. When the mobile telephone user leaves the call, remaining participants may renegotiate session features to activate video communications.
  
  
• Call Participant Management: During a session, a participant can bring other users into the call or transfer, hold, or cancel connections.

SIP Protocol Components

SIP has two basic components: the SIP user agent and the SIP network server. The user agent is effectively the end system component for the call and the SIP network server is the network device that handles the signaling associated with multiple calls. The SIP user agent consists of the user agent client (UAC) which initiates calls and the user agent server (UAS) which answers calls. This architecture allows peer-to-peer calls to be made using a client-server protocol.

The SIP network server element consists of three forms of server: the SIP stateful proxy server, the SIP stateless proxy server, and the SIP redirect server. The main function of the SIP servers is to provide name resolution and user location, as callers are unlikely to recall the IP address or host name of called parties. Using an easier-to-remember e-mail-like address, the caller's user agent can identify a specific server (or server cluster) to resolve called party address information.

SIP provides its own reliable transfer mechanism independent of the packet layer. For this reason, SIP does not require the services of the sigtran SCTP protocol and functions reliably over an "unreliable" datagram protocol like UDP.

SIP-T

SIP-T (SIP for telephones) is a mechanism that allows SIP to be used for ISUP call setup between SS7-based public switched telephone networks and SIP-based IP telephony networks. SIP-T carries an ISUP message payload in the body of a SIP message. The SIP header carries translated ISUP routing information. SIP-T also specifies the use of the SIP INFO method for effecting in-call ISUP signaling in IP networks.

PINT and SPIRITS

Sigtran is not the only IETF Working Group involved in defining new protocols to enable the integration of the PSTN with IP networks. PINT (PSTN and Internet Interworking) and SPIRITS (Service in the PSTN/IN Requesting Internet Service) are two IETF Working Group recommendations that address the need to interwork telephony services between the PSTN and the Internet. PINT deals with services originating from an IP network; SPIRITS deals with services originating from the PSTN.

In PINT, PSTN network services are triggered by IP requests. A SIP Java client embedded in a Java servlet on a web server launches requests to initiate voice calls on the PSTN. The current focus of this initiative is to allow web access to voice content and enable click-to-dial/fax services. In SPIRITS, IP network services are triggered by PSTN requests. SPIRITS is primarily concerned with Internet call-waiting, Internet caller-ID delivery and Internet call-forwarding.

ENUM

The IETF's ENUM (Telephone Number Resolution) Working group is devising a scheme to map E.164 telephone numbers to IP addresses using the Internet DNS (Domain Name System) so that any application, including SIP, can discover resources associated with a unique phone number. A SIP phone or proxy server would use number domain translation and DNS resolution to discover a DNS resource that would yield a SIP address at which a dialed number could be reached.

TRIP

The IPTEL working group is developing TRIP (Telephony Routing over IP), a policy-driven inter-administrative domain protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the routes to those destinations. TRIP is designed to allow service providers to exchange routing information in order to avoid over-provisioning or duplication of gateways using established Internet protocols.

If a telephone number does not have an associated SIP resource, the IP network routes the call to a telephone routing gateway, which connects to the PSTN. In an interconnect environment with many peering relationships between service providers, resources in the IP network need to discover which telephone numbers are associated with which gateways.

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