Session Initiation Protocol (SIP)

• At present, two standard systems are available for the interworking between softswitch Session Initiation Protocol (SIP) domain and traditional Public Switched Telephone Network (PSTN), SIP for Telephones (SIP-T) protocol suite of the Internet Engineering Task Force (IETF) and SIP with Encapsulated ISUP (SIP-I) protocol suite of the ITU-T. In this article, the SIP is introduced first because the two protocol suites are all extensions to it.

SIP

• SIP is a set of signaling protocols for establishing, modifying and terminating IP sessions, including multimedia conference over IP.

1. Functions of SIP
   • Name translation and user location: SIP uses a set of naming mechanisms similar to the Email address, to locate a specific called party wherever the party is. Each user is determined by all hierarchical ULR elements, for example, SIP: user@company.com, 80000001@beijing, or username@202.112.10.4. and can even be associated with the Email address directly.
   • Session parameter feature negotiation: SIP allows all the participants in a call to negotiate parameter characteristics of the session. For example, the videophone function cannot be used when several videophone users have a session with a mobile phone user. After the mobile phone user quit the session, the videophone users can renegotiate about using the video function.
   • Call participant management: During one session, the participants can invite other users to take part in, transfer, keep and cancel the connection.
2. Components of SIP
   • SIP consists of two basic components: SIP user agent and SIP network server, where SIP communications are carried out, as shown in Fig.1.
   • Database containing the user´s URL and location information)
   • The SIP user agent is actually the terminal system component of the call. The SIP network server is the network equipment that processes signaling related to multi-point call (IP conference), and it is responsible for resolving user names and locating users.
3. Applications of SIP
   • SIP supports powerful multimedia services and is extendable to support different applications. In fixed-line softswitch applications, SIP is located at the call control layer of the flat architecture, and supports call connection between different softswitches. From the viewpoint of routing, there are two cases for the usage of SIP architecture:
     • In the first case, normal ISDN User Protocol (ISUP) messages are encapsulated into SIP messages (with additional information added) for transmission. Functions such as call server, number, route analysis, signaling and service interworking remain unchanged, and route analysis routes the messages to the destination IP addresses.
     • The second case is based on the ENUM (telephone number mapping working group of IETF) database. In this case, the call control of the call server does not provide number and route analysis functions, which is absolutely different from that of current circuit-switched network. But service mapping and interworking are still required. Because the Circuit Identification Code (CIC), ISUP management process and Message Transfer Protocol (MTP) are not used, the standard ISUP needs to be modified correspondingly. #*Therefore, the network management is simplified (such as, there is no signaling network and routing definition). Also in this mode, the control of telecom carriers on networks is reduced and control modes have great changes, compared with current networks.
   • The aforesaid analysis shows that the application of SIP disables some functions of current telephone networks. To introduce these functions, you need to extend SIP.