VoIP is the transmission of voice over packet-switched networks.
Traditional voice networks utilised circuit-switching technologies. Effectively, a pair of copper wires is used to connect one party to the other, completing an electrical "circuit". With packet-switching, traditionally used by IP based networks such as the internet to transmit data, voice signals are broken down into tiny "packets" of digital sampled data, sent and then reassembled at the receiving end. The efficiency in voice over IP is in part down to this packet process. Multiple conversations (as many as 6,000) can be transmitted over one pair of copper wires.
Wider adoption of VoIP has been largely down to advances in IP networking technology, and processor speed increases. The faster a processor can sample voice data into small packets, and the faster those packets can be sent over an IP network, the better and more efficient VoIP becomes.
Due to the explosion of broadband in recent years, more and more people have fast internet connections at their disposal. With VoIP, every call you make can be treated as a "local" call - the internet has no concept of international. When you access a web page in the United States from the United Kingdom, you don't pay an "international" rate - you simply have a flat rate connection fee.
VoIP can be viewed as a similar utility to web access or email - if you're on the internet with a fast enough connection, you can speak to another party without incurring any call charges whatsoever, as long as the other party is using a device compatible with yours.
To ensure that two parties can talk to each other using Voice over IP, they need to ensure they can reach each other and are using two compatible clients. A VoIP client is simply an audio device that performs the dialing, sampling and packet sending. VoIP clients come in both hardware and software platforms, the software devices using your PC to do the work and usually used with an audio headset.
In terms of reaching each other, both parties VoIP client need to be talking the same language - using the same protocol.
SIP is now the most widely adopted protocol by the telco networks (with Skype as the only exception of note). VoIP User bases all its' services on SIP as do the larger commercial providers such as Vonage and Sipgate
For more information on SIP see SIP, RTP and NAT
H.323 is now largely redundant, although still used as a backbone interconnect between traditional telco branch exhanges.
IAX2 has been widely adopted as the internal PBX protocol of choice as a result of the success of asterisk, an open-source PBX application.
While cost-saving is often looked at as the fundamental reason for choosing it as a telephony system, having voice broken down into packets for transmission comes with other advantages. Routing and re-routing of call data is easier, and quicker. Follow-me type clients are easily implemented - wherever you have an IP address (or internet connection) you can receive calls on your number.
On the downside, packet-switched networks have not yet reached the level of redundancy of the PSTN (Public Switched Telephone Network). Reliability of your telephony with VoIP wholly relies on the reliability of your internet connection.
There is a system in place, ENUM, for the translation of traffic destined for a PSTN telephone number to an alternative internet route. The organisation and heirarchy of ENUM as a public service remains a work in progress for the telecoms community.
VoIP User was created as a community funded network of users who want to test and experiment with IP telephony. Our services are limited to what we are financially capable of within that remit.
Our servers and PSTN gateway are funded by the use of our non-geographic DID (DIrect Dial) inbound numbers. The revenue we generate on your use of those numbers goes straight into funding our servers and outbound PSTN routing. VoIP User itself is non-profit - we recycle all revenue into user services.
We run several servers - the one running this website, our SIP server (on a dedicated host in Paris, France) and an Asterisk server which we use for additional test facilities such as our echo test server.
Our entire network is based on the SIP protocol, and we have peering arrangements with other SIP based telcos such as Gossiptel, Plus Net and others on the way. Calls can be made from clients connected to our server, to users on those peered networks without a cost to the community.
Calls can also be made to the PSTN (regular landline phone lines) within certain restrictions and guidelines. Calls to the PSTN have to be paid for by us, and we do that using the revenue generated by the community in our outbound "pot".
We operate primarily on a basis of trust. In order to keep abuse off our network, we have developed an algorithm which we run on our server, which gives each user a dynamic rating based on a scale of -10 to +10. This User Rating is calculated automatically on our server based on calls made, calls received and general contribution to the community including users writing reviews and helping other users on the forum.
The VoIP Wiki - member maintained resource dedicated to VoIP
Session Initiation Protocol (SIP) - a site about SIP maintained by it's creator, Henning Schulzrinne
The H323 Project - An open source implementation of the protocol
(c) VoIP User 2005