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SIP Stack


SIP is a communique protocol created via IETF and distinctive in RFC 3261. It allows the establishment, management, and termination of Internet telephone calls, video conferences, and multimedia connections. The SIP stack is important for enforcing SIP in Solaris OS and includes numerous operational components, every playing a unique function.

SIP Stack

Here's a quick assessment of these additives:

  • Header Management Layer: This layer provides the interfaces required to build, parse, examine, and validate SIP headers.
  • Transaction Management Layer: SIP is based on a request/response transaction model like HTTP. This layer handles application-level retransmissions, matching responses to requests, and application-level timeouts.
  • Dialog Management Layer: A dialog is a persistent peer-to-peer SIP relationship between two user agents. This layer holds state information, which can be used to construct a request within a session.
  • Message Formatting Layer: This layer constructs a SIP message for an incoming message before delivering the message to the application. For outbound messages, it adds a Content-Length header followed by an empty line, meeting the requirements of RFC 3261.
  • Timer Management Layer: The SIP stack uses several timers. This layer provides timeout and un-timeout routines for these timers.
  • Connection Manager: This component provides input/output functionality. It is not a part of the SIP library but interacts with the library using well-defined interfaces.

Each of these components plays a vital role in ensuring the smooth operation of the SIP stack, enabling efficient multimedia communication sessions.


The Java APIs for Integrated Networks (JAIN) is a JCP work organization coping with telecommunication standards. Put Java and SIP collectively and you get the JAIN SIP API, a trendy and powerful API for telecommunications. This API is generally used for patron-side utility improvement. Other box-based technology, like SIP Servlet API, are better desirable for server-side improvement.

SIP Stack


To get the JAIN SIP API libraries, go to the Jain-sip project home page. You'll need to get these files:

  • SIP interfaces and main classes
  • SIP reference implementation
  • Logging service
  • Concurrency Utilities

SIP Tutorial

SIP is a signalling protocol designed to create, regulate, and terminate a multimedia consultation over the Internet Protocol. It is an utility layer protocol that incorporates many factors of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). This educational covers most of the topics required for a fundamental understanding of SIP and to get a feel of how it really works.

SIP and Telecom Testing

This tutorial has been prepared for specialists desiring to analyze the basics of SIP and make a career in telecom testing. Before intending with this academic, you need to have an excellent grasp of initial networking standards which include a number of the fundamental protocols which include TCP, UDP, HTTP, SMTP, and VoIP.

SIP Address

In traditional telephone verbal exchange, the sender and receiver are diagnosed by means of their respective cellphone numbers. However, in SIP (Session Initiation Protocol), either birthday party can be diagnosed using an e mail deal with, IP cope with, or cellphone number. These identifiers are represented as URLs the use of the SIP scheme.

SIP Messages

SIP is a textual content-based protocol modelled on HTTP. It uses messages in ASCII text. Each message has a header and a body. The numerous SIP messages consist of INVITE (requests for initiation of a consultation), ACK (confirms that session has initiated), BYE (requests for the termination of the consultation), OPTIONS (queries a host approximately its abilities), CANCEL (cancels the pending request), and REGISTER (informs a redirection server approximately the consumer's modern location).

SIP Session

A simple session using SIP consists of establishing a session, communication, and terminating the session. The session establishment requires a three-way handshake. The caller sends an INVITE message. If the callee is willing to start, he/she sends a reply message. To confirm that a reply code is received, the caller sends an ACK message. After the establishment of the session, the caller and callee communicate using two temporary ports. The session can be terminated by using a BYE message sent by either the caller or the callee.


In the following examples, Alice calls Bob using his SIP URI, 'sip:[email protected]'. Bob answers Alice with a successful Response. The message is an example of an INVITE request containing an SDP message being responded to with a "200" OK response.


SIP is a powerful protocol that allows for establishing, managing, and terminating multimedia connections. With its flexibility and wide range of applications, it's an essential tool in the world of telecommunications.

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