There are a couple of different ways you can make VoIP (Voice over Internet Protocol) calls. You can use a software package (or ‘Soft Phone') which then allows you to connect to landline phones or other computers, or you can use your existing telephone equipment though a VoIP service provider.
It's often good to test out VoIP using a free package suck as Skype or Gizmo. You don't need to sign a contract or buy any extra equipment, you just need a sound card and a headset with microphone and headphones. It's also possible to use an internet telephone, which you plus into the USB port or sound card of your computer.
If you're interested in VoIP, you won't be struggling for choice. There are more than 50 companies offering their own version of VoIP. The range varies from VoIP that only works on a specific computer platform, such as Linux, right through to offerings that can be used on multiple types of computers and operating systems. Using these programs allows you to make free computer-to-computer calls, but if you use them to connect into the regular phone network (i. e. to call someone on their phone) you will usually have to pay a small fee.
It wasn't that long ago that if you wanted to talk to someone else via VoIP, they had to have the same software installed as you did. This was very restrictive, and led to the development of the SIP (Session Initiation Protocol) standard. Basically, SIP allows any software to connect computer-to-computer. Not all VoIP services use it, and Skype is one of the biggest VoIP providers that has a proprietary protocol that still can't link in to other types of software. Nearly all VoIP packages, however, can be used to call landline or cellular phones.
As long as you have a broadband internet connection, you can use your soft phone. It's easy to call a friend in Asia or a business associate in the next office, as long as you both have appropriate software installed.
All VoIP software packages perform similar functions, even though they have their own unique interface to do it. To contact another person, it's usually as simple as typing in their user name or number. If they're online, a box will popup to let them know you're calling. They can see who's calling and decide whether to accept or reject the call.
While the VoIP software is waiting for the call to connect, it collects information about the speed of the connection, and what type of codec is being used on both computers to compress and decompress audio data. Once the call connects, the two computers then have to agree which type of codec will be used for the best possible result based on the connection speed.
When you talk, your voice is picked up as an analog signal. When you speak into your microphone or telephone, the analog signal is divided into small steps, and given a numerical value. This is similar to the technology used by CDs, which convert analog signals into digital data by sampling sound 44,100 times per second. Now the digital data is compressed, and broken up into packets for transmission over the internet. Information about the data is encoded into each packet, including its origin and destination. Not all data packets travel via the same data stream, so it's important to identify each one individually.
The data travels along routers to its destination. As each one is issued, your service provider rapidly determines which router to send it along, based on current usage patterns. There are thousands of routers involved in making up the internet, so your data packets can travel along a large number of different paths, depending on which one is the fastest when it's sent. The data packets travel from router to router until they reach their destination.
As a result, the data packets don't generally arrive in the same order they were sent, so at the other end the VoIP software will examine the time stamp on each data packet and reassemble them in the correct order. Sometimes, if one packet takes too long, it will be dropped, leading to a slight gap in the transmission. If this didn't happen, and a data packet had gone astray, there would be huge gaps in the transmission while the VoIP software waited for the missing packet to arrive.
The quality of the connection is determined by the speed of the internet connection at either end and also the general speed and condition of the various routers the data travels through. This quality determines how many packets are dropped, and so the final quality of the transmission.
Once the data is received, the digital information is converted back into an analog signal, using the Analog to Digital Convert in the sound card or telephone set.
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