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Conventional telephony
 When you make a telephone call, the telephone exchange establishes an exclusive connection to the number you dial. While you're having the conversation, anybody else trying to dial either party will get an 'engaged' tone. That's
essentially what a circuit-switched network does. It establishes an exclusive and continuous physical connection between two parties. Circuit-switched technology itself has evolved quite a bit. It started with cordboard switches where an operator manually connected two parties through a cord. From there, it moved on to SxS systems, also called the Strowger exchanges, named after its inventor. These were electromechanical in nature. After this came the crossbar exchanges, which are still being used in many countries. Crossbar used electromagnetic principles. Today, electronic telephone exchanges are fairly common, which are more compact and powerful, and convert voice into digital signals for transmission.
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Why is Internet telephony illegal? |
| If one were to get philosophical, there are two types of illegalities: those of the moral kind and those of the economic kind. While murder and stealing fall in the first category, Internet telephony clearly does not. So, it could be an economic offence at best. Offence against whom? Is the country losing anything because you use Internet telephony? Is some other country benefiting at our cost if you switch over to Internet telephony? Clearly not.
The only loser is VSNL/BSNL whose revenues would be affected, as the bulk of their revenue still comes from voice traffic, and they seem to be in no hurry to attempt a switch over to a different way of life. No wonder then that Internet telephony (and VoIP) is treated as illegal.
But clearly, the day is not far off when the ban can no longer hold. Any private player with a voice and an ISP license could open the doors for VoIP and Net telephony, and change the way we use the telephone or the PC, and the way we pay for it, once and for all. |
Digitizing sound is an interesting process. When we make a telephone call, our voice is first converted into analog electrical signals. This signal is then encoded into digital format using a technique called PCM (Pulse Code Modulation). This technique takes samples of the analog signals at a rate of 8,000 samples per second. Each sample therefore represents 125 microseconds of a voice stream, and is eight bits, or one byte long. This signal is then carried over high-speed digital lines and again decoded into an analog electrical signal at the receiving end. The analog signal is finally converted into the original sound.
Speaking of sound, any conversation consists of two components-sound and silence. When the digital sound signals are transmitted over a circuit switched network, both components have to be sent. Not only that, but the order of transmitting signals also has to be retained, else quality of transmission suffers. That's why all equipment in a circuit-switched network must be highly synchronized using expensive TDM (Time Division Multiplexing) equipment. Since sound and silence are both transmitted, a lot of bandwidth gets wasted in circuit-switched networks. In fact, one voice conversation requires a 64 kbps channel, which is quite a lot of bandwidth.
How VoIP works
In VoIP, analog voice signal is digitized using PCM. These digital voice samples are then buffered on an IP gateway. This device converts the PCM data stream into a compressed IP packet stream using DSPs (Digital Signal Processors). DSPs are responsible for converting from analog to digital as well as compression. The set of PCM samples are analyzed as a discrete set of binary data. It checks the speech for all the moments of silence, which are a lot. Even when we speak, there are pauses in between that go unnoticed to the human ear, but are quite discernible to the sampling device. The length and beginning of these pauses is noted, while the remaining 'silence' is removed from the data set. Similarly, redundant data is also removed, making the data set more compact. Finally, an IP header is attached to this compressed data, which is then sent out on the network as discrete data packets.
Once the voice packet is sent out, it finds its way to the destination just like any other data packet. It passes through various routers and switches to reach the destination gateway. Here, it gets decompressed, meaning all the periods of silence and redundant data are reinserted, and is finally decoded to produce an approximation of the original sound. The compression algorithms used in this process can compress the voice signals and can even carry voice over as little as 5.3 kbps bandwidth.
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