This part is for technical people. It is very nice to know what VoIP can do for you and why you should or should not go your stuff VoIP, but if you want to get deep into the subject, you need to understand some technicalities, which are briefly introduced here.

What is done in a VoIP connection?

To setup a VoIP communication system, we need to have the following things done:

1. First, an ADC (analogue to digital converter) is needed to convert analogue voice to digital signals (bits).
   
2. Now the bits have to be compressed in a good format for transmission: there is a number of protocols that we'll see after.
   
3. Here we have to insert our voice packets in data packets using a real-time protocol (typically RTP over UDP over IP).
   
4. We need a signalling protocol to call users: ITU-T H323 does that.
   
5. At RX we have to disassemble packets, extract data, then convert them to analogue voice signals and send them to sound card (or phone).
   
6. All that must be done in a real time fashion because we cannot be waiting for too long for a vocal answer!
   

The Analog to Digital Conversion

This is made by hardware, typically by card integrated ADC. Today, every sound card allows you to convert with 16 bit a band of 22050 Hz (for sampling it you need a frequency of 44100 Hz for Nyquist Principle) obtaining a throughput of 2 bytes * 44100 (samples per second) = 88200 Bytes/second, 176.4 KBytes/s for stereo stream.
For VoIP we don't need such a throughput (176kBytes/s) to send voice packet. We will see later other codings used for it.

Compression Algorithms

Now that we have digital data, we may convert it to a standard format that could be quickly transmitted. The standard format not only serves the purpose of standardization, and hence broad acceptance, but also reduces the bulk of transmitted data through compression, thus increasing the speed. Compression in itself, in some limited way, is a means of encryption, which may prove important in certain cases. Below is a list of protocols used for compression.

• PCM, Pulse Code Modulation, Standard ITU-T G.711
  
  • Voice bandwidth is 4 kHz, so sampling bandwidth has to be 8 kHz
    
  • Each sample is represented with 8 bit (having 256 possible values)
    
  • Throughput is 8000 Hz *8 bit = 64 Kbit/s, as a typical digital phone line
    
  • In real application mu-law (North America) and a-law (Europe) variants are used which code analogue signal a logarithmic scale using 12 or 13 bits instead of 8 bits
    
  
  
• ADPCM, Adaptive differential PCM, Standard ITU-T G.726
  It converts only the difference between the actual and the previous voice packet requiring 32 kbps (see Standard ITU-T G.726).
  
  
• LD-CELP, Standard ITU-T G.728
  
  
• CS-ACELP, Standard ITU-T G.729 and G.729a
  
  
• MP-MLQ, Standard ITU-T G.723.1, 6.3kbps, Truespeech
  
  
• ACELP, Standard ITU-T G.723.1, 5.3kbps, Truespeech
  
  
• LPC-10, able to reach 2.5 kbps
  
  

These last protocols are the most important because they can guarantee a very low minimal band using source coding; also G.723.1 codecs have a very high MOS (Mean Opinion Score, used to measure voice fidelity) but attention to elaboration performance required by them, up to 26 MIPS!