The Quick Start Guide to VOIP for the Home, Home Office, and Small Business

How VOIP Works—A Nuts and Bolts VOIP Tutorial

Previous:  [ VOIP Tutorial Part 1 ]

Step 3—Turn Voice into Data

VOIP is based on digital data transmission. So, when you make a VOIP call the first thing that happens is the analog signal of the your voice is converted to digital data. This voice-to-data conversion is done with an Analog-Terminal Adaptor (ATA).

The ATA can be a separate box that plugs into your phone and then into your broadband modem, or it can be built into a special IP phone (sometimes called VOIP phone or SIP phone). If you are using a computer to make calls, the ATA function is built into the VOIP software. Service providers, such as Vonage, will provide the ATA or VOIP phone to you as part of their VOIP package.

The ATA divides the analog voice signal into chunks—or packets—and then compresses the chunks which significantly reduces the amount of digital data to be transmitted. Voice compression is handled by a CODEC (Coder/Decoder).

Now that your voice has been converted to digital and compressed, it can be sent over the Internet. The data stream must be divided into packets which, besides containing the voice data, also has information concerning its destination and their sequence in the data stream. The transmission of voice packets over the internet are handled by various protocols.

Among other things, protocols contain information about the sequence of the data packets so they can be reconstructed in the correct order at their destination. In traveling from A to B, some packets may be dropped. However, there is usually enough information to make the conversation legible. The number of packets that are dropped depends on the speed of your Internet connection in the distance between the two parties.

Step 4—Turn Data Back into Voice

Before voice data packets can be received by a listener, they must be reassembled in the correct order and converted back to analog. If you are making a computer to computer call, the voice data stream is reassembled at the receiving party's computer by the VOIP software. The entire conversation never leaves the IP world--the Internet.

If you are calling someone on a landline, the voice data stream jumps off the internet and onto the regular phone lines at a VOIP gateway. The gateway is usually located in the city or region where the receiving party is located. This is the reason why long distance rates using VOIP can be so cheap. VOIP service providers basically have their VOIP gateways set up in major centers all over the world. When you call someone in one of these cities, all you are really paying for is the cost of a local call from the VOIP gateway to the receiving party.

Step 5—Talk!

Now that your phone call has been routed through the Internet, and possibly shunted through a VOIP gateway onto the PSTN, and has made it to the receiving party (all in a matter of seconds), you can get talking!

In many cases the person you are calling won't even know that this is a VOIP call and the voice quality will be the same as on a regular landline call. However, if too many voice packets were dropped along the way, or the Internet connection is slow or poor, you may notice any of the following:

  • You or the other party's voice is breaking up, or words seem to drop out (packet loss)
  • A time delay between the speaking and hearing  (packet delay or lag)
  • Noisy lines or background noise, and voice echo (traditional phone line problems if the call has been integrated with the PSTN)

 

Next: [ Deciding to Use VOIP—Weighing the Pros and Cons ]

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