Introduction to SIP Protocol

SIP is a tool for setting up Internet-based communication, the enpoint of which can be a PDA, a smartphone, a tablet or a computer with microphone and loudspeaker or headset or webcam. If you have all these with the necessary applications, you can make audio and video calls through the Internet. This page will explain you how, so if you have heard about these, but you have no idea what they mean, or if you do, but you wish to learn more about it, then please read and watch.

What is a Protocol?

Certain regulations are needed for the data to be transmitted via the Internet. These sets of regulations are called protocols. According to this, data get compressed into smaller data packets before they are sent to the appropriate IP address (Internet Protocol Address is given to each computer in a network to provide a destination address to data packets).

What is a SIP protocol?

SIP is one such protocol, Session Initiation Protocol, to be more exact. As its name suggests, it initiates sessions on the Internet. It builds up and terminates the line, and gives room for parameter changes in mid-session.

Parts of SIP

  • UA (user agent): the device that uses SIP, like an IP Phone (a device that converts voice into digital data by itself), a computer or a conference bridge.
  • Redirect server: for requests, it returns new locations.
  • Proxies (stateless and forking): they both route call requests.
  • Registrar: name and addressee collection is kept by this.

Services

  • Voice mail: User Agent with a special URL (Uniform Resource Locator, the global address on the World Wide Web) and RTSP (Real Time Streaming Protocol controls the transfer of real-time media data).
  • DTMF (Dual-Tone Multi-Frequency) carriage: PSTN (Public Switched Telephone Network, or the old, conventional way of communication) numbers are carried by it, in Real-time Transport Protocols (RTP delivers audio and video through the Internet in a standardised packet format).
  • Calling Card and a voice server
  • Call forwarding: patching through a caller to another extension without answer
  • Call transfer: having picked up the phone, you patch through a caller to another extension.
  • Call hold: you have the caller waiting before somebody answers
  • Caller ID: usually a number, and it also provides extensions.

SIP can:

  • Start and Terminate VOIP calls and multimedia conferences
  • Give you notification of your events
  • Send general and text messages
  • Signal transport.

Your communication system can include SIP phones, VoIP phones, softphones, computers, smartphones and so on. Ozeki Phone System will always be able to connect your devices in a network, in which they all can keep contact with each other and everybody else in the world, unless you would like to limit it to certain areas. With Ozeki Phone System XE you will be able to call any device with an audio input and output device, since Ozeki Phone System XE provides the highest quality and highest expertise at the next generation of communiation.

Read the following pages for further information:

For a better understanding, please watch our video:

Introduction to SIP Protocol (Video tutorial)

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