SUBSCRIBE and NOTIFY methods

The SIP specification has been extended over time to support a general mechanism allowing for subscription to asynchronous events. Such events can include SIP proxy statistics changes, presence information, session changes and so on.

A user agent interested in event notification sends a SUBSCRIBE message to an SIP server. The SUBSCRIBE message establishes a dialog and is immediately followed by the server replying with 200 OK response. At this point the dialog is established. The server sends a NOTIFY request to the user every time the event to which the user subscribed changes. NOTIFY messages are sent within the dialog established by the SUBSCRIBE.

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SIP INVITE method

Calls are started by means of the methods INVITE together with SDP (Session Description Protocol) which carry the information necessary to allow the endpoints of the calls to exchange audio in form of RTP (Real Time Protocol) packets.

Let’s see a typical call dialog: Continue reading

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Understanding REGISTER method

SIP is a peer-to-peer protocol where the roles client – server and exchangeable depending on who starts a session. In reality most deployments foresee a process called registration (method: REGISTER) which allows a central server (registrar) to store the location of a SIP User-Agent.

A SIP Phone is a client to the central Unified Communication Platform (registrar) – and the UC platform is a client to the SIP Server of the operator (registrar).

Once the SIP Server gets to know the location of an SIP Client, it can deliver calls and other messages to it from other Clients connected to the same Server.

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Analyzing SIP and SIP calls 

This blog article will be dedicated to SIP and SIP calls. It is meant to provide an introduction for UC Engineers to the main headers and guide through the debugging of the most popular call scenarios that will be described in the next articles.

As we have seen in SIP and XMPP standards in Unified Communications and Media transfer in Unified Communications – SDP protocol, SIP (Session Initiation Protocol) and SDP-RTP have become the de facto replacement mode for traditional analog and digital lines provided by operators all over the world.

SIP is also the most popular signaling mode to handle VoIP calls.

Being able to understand SIP sessions has become for a Unified Communication Engineer as important as in the past for Telephony Experts to read signaling traces in Q.931 / Q.921 / DS1 / QSIG generated by T1 / E1 PRI – BRI ISDN lines.

Before Starting the Analysis

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