Making a VoIP Call - How VoIP soft phones work

The other party can see who is calling and can either accept or reject the call. Before the popup appears, however, there has already been some communication between the two computers. The VoIP software has information about the speed of your Internet connection and the type of codec that can be used to compress and decompress audio data. When a call request is made, both computers have to negotiate which codec is going to be used to make allowances for the connection speed.

The first step in making computer-to-computer telephone calls is to convert your voice into digital data. As you speak into the microphone of the headset or telephone set connected to your computer, it is 'sampled' -- converted to digital numbers by dividing the analog signal into individual steps, each of which are given a numerical value. This is the same technology behind audio CDs which convert analog signals into digital data by sampling the sound 44,100 times per second.

CD-quality sound, however, is not needed for Internet telephony. Voice data can be compressed substantially and still remain understandable. The compressed voice data is encapsulated in data packets which will be sent over the Internet. The destination of the data is encoded in each packet, but the route each packet takes may be completely different from other packets in the same data stream.

The Internet is made up of thousands of 'Routers' which are responsible for delivering data in an efficient manner. Routers have information about the data load of other routers in the network and can use this information to determine the fastest path. The router examines the destination address of each packet and forwards it to the next router on the path. In this manner, the data packet is forwarded from router to router until it reaches its destination.

Since the conditions of data paths along the Internet are constantly changing the most efficient path for one data packet may not be the same for the next packet. This means that VoIP data will probably not arrive at its destination in the same order that it was sent. The data can be reconstructed in the proper order because each packet has a time stamp on it, but in order to minimize the delay between one person speaking and the other person hearing the voice, some of the packets may have to be dropped.

The quality of the connection depends in part on how many packets are dropped. This in turn depends on the speed of the Internet connection at each end and the general condition of the Internet pathways.

Once the data has been received it is converted back into an analog voice signal with the Analog to Digital Converter (ADC) on the sound card or telephone set.

Try out VoIP for free! All you need is a headset attached to the sound card of your computer. Next, download one of the many VoIP software packages. Skype, Gizmo, Free World Dialup, and Net2Phone are some of the big names. With the software installed, invite all your friends to download the same software and when someone is up and running, give them a call!

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