SIP Protocol Fundamentals | SIP and VoIP
SIP full form is Session Initiation Protocol. A signalling protocol, broadly used for establishing, connecting and disconnecting communication paths, sometimes voice or video calls over the Internet Protocol. SIP is a standardized protocol with its foundation coming from the IP neighbourhood and typically makes use of UDP or TCP. The specification is given in RFC 3261. The protocol can be utilized for establishing, modifying, and terminating two-party (unicast) or multiparty (multicast) communication channels, consisting of a number of media streams. Modifications can embody altering IP addresses or/or ports, inviting extra individuals, and including or deleting the media streams.
As per the OSI model, SIP is an example of an application layer protocol. Being a signalling protocol (like SS7). It only carries the media and other information. Actual media is outside of the scope of the protocol. RTP is another protocol for sampling and carrying media over a session set up by the SIP protocol.
The following are the basic function SIP does for communication over IP.
- User Registration – A end IP phone of a VoIP application registers itself to send or receive voice or media sessions.
- User Capabilities – Determining the end-user parameters for a call or session.
- Setup of a session – Does the initial level of signalling for setting up a session. When an IP phones dial a number, session setup procedure.
- Session management or update – After a successful session setup, there may be a change in session properties. Session management may also include call transfer.
- Session Termination – In the end, all media-related resources free gracefully.
SIP Network elements.
The SIP network has nodes for registering a SIP client to the network and routing the calls. The following network elements are in the VoIP network.
User Agent – This is the application of an IP phone, that initiates or termination a call.
Registrar – It is a logical server. A user agent registers itself with a registrar in the network. Using the server, a user agent can be located. When there is an MT call to the SIP user agents, the registrar provides the serving IP of the agent.
SIP proxy – Routes the call messages and others. Generally, a REGISTRAR is co-located with the proxy.