Wednesday, November 18, 2009

What is a SIP server?

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar.

Although the SIP server is the most important part of the SIP based phone system, it only handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the media server in RTP.

An example of a SIP server is 3CX Phone System - a Windows-based SIP server

What is a SIP-URI?

A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user’s SIP phone number. The SIP URI resembles an e-mail address and is written in the following format:

SIP URI = sip:x@y:Port
Where x=Username and y=host (domain or IP)

Examples:
sip:joe.bloggs@212.123.1.213
sip:support@phonesystem.3cx.com
sip:22444032@phonesystem.3cx.com

The SIP URI standard has been defined in the RFC 3261 standard. 3CX Phone System for Windows uses SIP URIs

Links:
RFC 2396 - Uniform Resource Identifiers (URI): Generic Syntax.

What is a STUN Server?

A STUN (Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. computers behind a firewall) to setup phone calls to a VOIP provider hosted outside of the local network.

The STUN server allows clients to find out their public address, the type of NAT they are behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between the client and the VOIP provider and so establish a call. The STUN protocol is defined in RFC 3489.

The STUN server is contacted on UDP port 3478, however the server will hint clients to perform tests on alternate IP and port number too (STUN servers have two IP addresses). The RFC states that this port and IP are arbitrary.

More information about STUN and VoIP in general can be found in our SIP / VoIP Video tutorials, 'Voip Nuggets'. VoIP Nuggets are short youtube technical training tutorials about VoIP. Click here for the latest list of VoIP Nuggets

Stun functionality is seamlessly handled by 3CX Phone System for Windows - an easy to install windows based PBX.

Tuesday, November 17, 2009

What is Unified Communications?

Unified Communications is defined as the process in which all means of communication, communication devices and media are integrated, allowing users to be in touch with anyone, wherever they are, and in real time.

The objective of Unified Communications is to optimize business procedures and boost human communications by simplifying processes.

Read more about Unified Communications.

What are the benefits of an IP PBX?

  • Much easier to install & configure than a proprietary phone system
  • Easier to manage because of web based configuration interface
  • No need for separate phone wiring
  • Allows users to hot plug their phone anywhere in the office - users simply take their phone, plug it into the nearest ethernet port and keep their existing number!
  • Allows easy roaming - calls can be diverted anywhere in the world because of the SIP protocol characteristics
  • Significant cost reduction by leveraging Internet
  • SIP standard eliminates proprietary, expensive phones
  • Scalable
  • Better reporting
  • Better overview of system status and calls
  • More IP PBX benefits

What are the benefits of an IP PBX?

  • Much easier to install & configure than a proprietary phone system
  • Easier to manage because of web based configuration interface
  • No need for separate phone wiring
  • Allows users to hot plug their phone anywhere in the office - users simply take their phone, plug it into the nearest ethernet port and keep their existing number!
  • Allows easy roaming - calls can be diverted anywhere in the world because of the SIP protocol characteristics
  • Significant cost reduction by leveraging Internet
  • SIP standard eliminates proprietary, expensive phones
  • Scalable
  • Better reporting
  • Better overview of system status and calls

What is H323?

H323 is a set of standards from the ITU-T, which defines a set of protocols to provide audio and visual communication over a computer network.

H323 is a relatively old protocol and is currently being superceded by SIP – Session Initiation Protocol. One of the advantages of SIP is that its much less complex and resembles the HTTP / SMTP protocols.

Therefore most VOIP equipment available today follows the SIP standard. Older VOIP equipment though would follow H 323.