Soft Phones – a New Level in Business VoIP Communication

Soft Phones – a New Level in Business VoIP Communication

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 (sometimes called a ‘SoftPhone’) that allows you to connect to other computers or 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 in with a specific VoIP provider. All you need is a sound card 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.

VoIP software seems to be the latest craze — there are at least 50 companies offering their own version of VoIP. Some of them are for specific computer platforms like Linux but others can be used on many kinds of 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 or also called POTS – Plain Old Telephone Service).

Up until recently, the major VoIP disadvantages (with computer-to-computer calls) was that both parties had to have the same kind of software installed in order to make a connection. The emerging SIP (Session Initiation Protocol) standard, however, allows any SIP software to connect. Some software does not use SIP – Skype, for example, uses a proprietary protocol and cannot connect to other types of software. Almost every software package, however, has the ability to connect with landline or cellular phones.

Softphones 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 who lives down the street as long as both have the proper software installed.

How It Works

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 username 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 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 a computer-to-computer telephone call 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 is 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. For example, the single word ‘Hello’ requires about 43 kB in CD-quality sound. Compression algorithms can bring that down to about 2 kB!

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 timestamp 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.