VoIP Technology

To establish a VoIP ‘tunnel’ on the Internet a few things have to take place.

1. First the analogue (voice) signal must be converted into a digital signal.
2. The digital signal is then compressed to make for more efficient transmission. There are several different formats/protocols that this compression can use.
3. The transmission of the digital information has to be done using a Real Time Protocol.
4. A signaling protocol is also used to call the users on either side of the conversation. This protocol is called ITU-T H323.
5. At the receiving side the packets of digital data are then uncompressed and converted back into analog voice signals that are then transmitted to the PCs sound card or the recipient’s telephone.
6. And all of this takes place in real time…

Analogue to Digital Conversion and VoIP Transmission

In today’s PCs all sound cards are capable of performing this step of converting an analog voice signal into digital data.

Next this digital data is compressed using one of a number of techniques.
The VoIP data packets are then encapsulated in RTP (Real-Time Transport Protocol) packets which are inside UDP-IP packets. This is because VoIP doesn't use TCP (because it is too heavy for real time applications) and so instead UDP is used.

UDP has no control over the order in which packets arrive at the destination or how long it takes them to get there. Both of these are however critical to overall voice quality in terms of how well you can understand what the other person is saying, and conversation quality in terms of how easy it is to carry out the conversation in the first place! The Real Time Protocol though solves this problem by enabling the receiver to put the packets of data back into the correct order without having to wait too long for any packets that may have either lost their way or are simply taking too long to arrive. Note that not every voice data packet is needed, but a continuous flow of many of them that are in the correct order is required.

All of this technology is lumped into a concept which we call QOS (Quality Of Service). This is a handy way to assess how well the VoIP technology is performing when it comes to delivering crystal clear real time phone conversations and is a key differentiating factor between VoIP services that work (with a high QOS), and those that are less reliable (with a low QOS).

MyCostSavingCall is a high QOS VoIP provider.