VOIP Compression  

First, VoIP is not illegal in India: read my lips: VoIP is not illegal, Internet telephony is. Now that I have got your attention... Internet telephony means connecting a pstn telephone to via internet to another speaker. For instance, rediff offers voice chat, and vsnl has no issues on this front, nor is there a problem with yahoo voice chat. Vsnl is object only to calling a regular phone number via the internet.

Okay, now to a technical issue. The first issue is of bandwidth. The audio is basically a waveform. The height of the wave (amplitude) is measured about 8000 times a second, each of these samples (8000 for every second) is a 16bit value. That means, there will be 16000 bytes every second. We will require 128kbps of data transfer in each direction to maintain a full duplex voice conversation. This is not possible in the indian scenario (or even in the west). 128 kbps is too much of bandwidth requirement, therefore, the need to compress voice. How do we do that?

The first way, is to map a range of 16 bit numbers to an 8 bit token value and transmit these 8 bit tokens, thus all the values within the range covered by each token is reduced to a single average value (loosing some clarity in the process). There are two mapping schemes in vogue - a law(used in europe) and u law (used in usa). This reduces the bandwidth requirements to 64kbps in each direction. Most of traditional telephony uses this scheme. It is fast, efficient, with hardly noticible degradation, but it wont work with a dialup modem (where 28.8 kpbs is what you can realistically expect in the indian conditions).

A second way of reducing bandwidth requirement is to make use of the nature of voice signals. Voice signals don't change rapidly in time, therefore two neighbouring values of audio signal change very little from each other. Therefore, if we know one signal value, we can only transmit the difference between the previous and the next value (using much less bits - typically 4 bits). This makes compression come down to about 32 kbps, still less for a modem but nearly there.

The third way is use something like GSM. GSM does an extremely complicated analysis of the voice - finds the principle distinguishing 'features' of the voice and just sends them over to the other side. It is like both sides having a joke book and one side tells the other 'joke number 52!' and both break out laughing. The trouble with these schemes is that they are all invariably patented. Therefore the problem of developing a VoIP solution for cheap. These schemes will work on as little as 4.8 kbps to give you a pretty good voice quality.

What is pertinent to us here is the inability to use any of the smarter ways of compressing voice without stepping over any patent. This work is very costly and laborious. No one will do it for free - unless it is sponsored by someone.

The other aspect is that bandwidth is only a matter of time. As soon as people move away from POTS line as the principle form of net access, the bandwidth restrictions will go away and 32kpbs codecs (discussed above as the second method) will become the dominent codec.

In either case, voip or internet telephony continues to be the killer app for telephony starved countries like india. and government should realise this and at least open it up in the rural areas.