Making a VOIP Call:

Part 1 -- Soft Phones

There are several ways to make VOIP calls. You can sign up with a VOIP service provider and use your existing telephone equipment, or you can use a software package on your computer (sometimes called a “Soft Phone”) that allows you to connect to other computers or to landline phones.

VOIP software such as Skype or Gizmo allows you to try out VOIP without investing in extra equipment or signing a contract that ties you to a specific VOIP provider. All you need is a sound card in your computer and a headset with a microphone and headphones. You could also use an Internet telephone that plugs into the sound card or USB port on your computer. 

Getting Set Up

VOIP software seems to be the latest craze--there are at least 50 companies offering their own versions. Some of them are for specific computer platforms, but others can be used on many different computers and operating systems. They allow you to make free computer-to-computer calls, but you have to pay a small fee if you wish to connect to the regular phone networks (PSTN: Public Switched Telephone Network, also called POTS: Plain Old Telephone Service).

Until recently, the major disadvantage of computer-to-computer calls was that both parties had to have the same kind of VOIP software. The emerging standard called SIP (Session Initiation Protocol), however, allows all SIP software to interconnect. Some software does not use SIP; Skype, for example, uses a proprietary protocol and can’t connect to other types of software. Almost every software package, though, has the ability to connect with landline or cellular phones.

Soft phones can be used anywhere in the world where a broadband Internet connection is available. You can call a business associate in Asia or your cousin Charley down the street, as long as both have the proper software installed. 

Now, For Your Call

Although each VOIP software package has its own unique interface, they are all similar in function. You usually call another person on the network by typing in their user name or number. If that person is online, they will see a pop-up box alerting them that you want to talk. The other party can see who is calling and can either accept or reject the call.

Before the pop-up appears, however, there has already been communication between the 2 computers. The VOIP software has information about the speed of your Internet connection and the type of codec (translator software) that can be used to compress and decompress audio data. When a call request is made, the 2 computers negotiate which codec will be used, depending on the connection speed.

Sound Into Data

The first step in making a computer-to-computer telephone call is to convert your voice into digital data. As you speak into the microphone, it is “sampled.” This means the analog signal is divided into individual steps, each of which is given a numerical value, thus being converted to digital data. This is the same technology behind audio CDs that 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. For example, the single word “Hello” requires about 43 kB in CD-quality sound. Compression algorithms can bring that down to about 2 kB, and it still sounds like “Hello”!

Routing For Speed

The compressed voice data is encapsulated into data packets to be sent over the Internet. The destination of the data is encoded in each packet, but the route 1 packet takes may be completely different from other packets in the same data stream.

The Internet is made up of 1000s of routers that are responsible for delivering data efficiently. 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 fastest 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 1 data packet may not work for the next. This means that VOIP data packets probably will not arrive at their destination in the same order they were sent. The data must then be reshuffled into the proper order (each packet has a time stamp on it), but to minimize the delay between 1 person speaking and the other person hearing the voice, some of the packets may have to be dropped.

Data To Sound

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 by the Analog-to-Digital Converter on the computer’s sound card or telephone set.