SIP Protocol Fundamentals | SIP and VoIP.

SIP full form is Session Initiation Protocol. It is a standardized protocol with its foundation in the IP neighborhood and typically uses UDP or TCP.

It is a control plane signaling protocol used in VoIP. In addition to providing procedures to establish, connect, and disconnect communication paths for voice or video calls over the Internet Protocol, the protocol also includes other major features such as messaging, call forwarding, and voice mail.

The specification is given in RFC 3261. The protocol can establish, modify, and terminate two-party (unicast) or multiparty (multicast) communication channels consisting of several media streams. Modifications can embody altering IP addresses and ports, inviting extra individuals, and including or deleting the media streams.

SIP protocol stack.

In accordance with the OSI model, SIP is considered an application layer protocol. It is a signaling protocol (similar to SS7) that conveys information about the media. It is outside the scope of the protocol to deal with actual media. For the purposes of sampling and carrying media for a session, RTP is used. 

SIP Protocol Stack

What ports does SIP use?

As an application layer protocol, SIP utilizes transport layer services. Port numbers 5060 and 5061 are used by default over TCP or UDP. The 5060 is used for plain text transfers, and the 5061 is used when TLS encryption is used.

What are the SIP protocol functionalities?

SIP performs the following functions for IP-based communication.

  • 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 signaling level for setting up a session? When an IP phones dial a number, the session setup procedure.
  • Session management or update – After a successful session setup, session properties may change. Session management may also include call transfer.
  • Session Termination – In the end, all media-related resources are 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.

SIP 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.