Well VoIP had a long and eventful journey before it became what it is today. Let’s first try to put together a bit of history to understand how it got to where it is today.
History
Well the first notable piece of history can be looked at as Thomas Edison’s invention of the phonograph. This rudimentary device allowed for a user to record his voice onto a piece of tin foil. The indentations made by this device could then be used to playback the recorded sound. The first form of telephony took place with the help of analog lines. Analog telephony used the properties of electricity to represent human speech. Apart from speech analog telephony used electricity to send signals across. These signals were used to indicate if a phone was busy or if dial-tone should be provided for example. An analog circuit consisted of a pair of wires one being the tip (connected to the ground) and the other being the ring (connected to the battery). A -48V DC current would be provided by the service provider side. These tip and ring combinations along with the combination of power being provided from the CO allowed for the phone to send signals. When the phone was placed on hook there would be an open circuit thus not allowing the current to flow through. This in turn would inform the central office that the phone is on hook. When the phone would go off hook the circuit would be connected and thus current would flow through indicating that the phone was off-hook. This form of signaling was known as Loopstart signaling.
There is another form of signaling known as Ground start signaling. Ground start signaling was formed to alleviate the issues caused by Loop start signaling. With loop start signaling there was an issue caused known as glare. Glare occurs when the user picks up his phone the same time an incoming call occurs causing the incoming call to be received by the wrong user. Ground start eradicated glare by grounding wires to signal for dial-tone.
Analog communications withstood the years but was eventually overcome by a need for a newer form. Analog signals while passing through a wire were subject to a loss in quality as they travelled large distances. To circumvent this issue repeaters were placed along the path however the repeaters would inadvertently amplify not only the voice but also the noise generated along the path. Another issue that came up was the number of wires that were required to be laid out by the service provider in order to connect all its users. It was time for analog to grow out of itself and evolve into something new. This came in the form of digital communication.
There is another form of signaling known as Ground start signaling. Ground start signaling was formed to alleviate the issues caused by Loop start signaling. With loop start signaling there was an issue caused known as glare. Glare occurs when the user picks up his phone the same time an incoming call occurs causing the incoming call to be received by the wrong user. Ground start eradicated glare by grounding wires to signal for dial-tone.
Analog communications withstood the years but was eventually overcome by a need for a newer form. Analog signals while passing through a wire were subject to a loss in quality as they travelled large distances. To circumvent this issue repeaters were placed along the path however the repeaters would inadvertently amplify not only the voice but also the noise generated along the path. Another issue that came up was the number of wires that were required to be laid out by the service provider in order to connect all its users. It was time for analog to grow out of itself and evolve into something new. This came in the form of digital communication.
Digital Communication
Digital communication involves the conversion of analog signals into an equivalent binary value. This binary value can then be sent through a TDM (Time-Division Multiplexing) circuit. TDM allowed multiple pieces of information to flow using the same single line. This eliminated the issues caused by analog communication. The number of wires reduced as a single line could be used instead of multiple lines. The signals being in binary were easily recreated at the receivers and thus eliminated the need for repeaters.
Digital communication can come in two cabling mechanisms T1 or E1. T1 is a cabling mechanism used in the United States, Canada and Japan. This T1 line allowed for up to 24 different channels of communication with a single wire. An E1 circuit is used in countries other than North America and Japan. An E1 line consists of 32 channels. These channels are called DS0’s (Digital Signal 0). Each of these DS0’s can transmit at 64 kbps. Each of these cabling mechanisms can transmit data using one of two methods CAS (Channel Associated Signaling) or CCS (Common Channel Signaling). CAS involves a procedure wherein signaling is sent along with the individual channels. In CCS a channel is dedicated purely for signaling.
CAS allows for the first 5 frames of each channel through as it is. The sixth frame has the last bit from each channel used for signaling. This would cause a slight degradation in voice quality but nothing too significant for the user to notice. In the case of a T1 CCS the 24th channel is used for signaling while in an E1 CCS the 16th channel is used for signaling.
Understanding Key systems and PBX
These T1/E1 lines were used with systems known as PBX (Private Branch eXchange) or key systems. These lines connected the systems with the PSTN (Public Switched Telephone Network) to make outbound calls across the world. These systems were similar to Central Office switches and basically connected multiple phones together within an enterprise environment. Key systems are a smaller implementation of a PBX. It supported fewer users and didn’t have all the features a PBX could offer. These systems were quite resilient and would have an uptime of 99.999% and a lifespan of 7 – 10 years. They were made of three components
Line cards – Used to connect phones present in the network with the PBX
Trunk cards – Used to connect one PBX to either the PSTN or another PBX
Control Complex – The brains of the whole operation. It decides on how calls are routed and setup.
Well we moved up from analog to digital communication and now let us understand why there was a need for a new more improved technology in the form of VoIP.
Benefits of VoIP
Cost – The first thing on everyone’s mind is cost. A factor by which every decision is made. And this is one of the most significant features of VoIP. Instead of paying the PSTN operator for calls between branches with VoIP you can send the call through the existing WAN connection between the sites. Reducing not only the call cost but also the need for any T1/E1 lines. MACs (Moves Adds and Changes) are significantly cheaper with a VoIP based system. Instead of running separate cables for both voice and data a single cable is required thus reducing cabling costs.
Portability – VoIP communications make use of softphones or even hard phones to allow you to literally take your phone with you. You can take your phone home and it would function the same as it did in the office.
Rich unified media – VoIP brings with an ecosystem where e-mail, conferencing, voicemail, voice and video all coexist in a single environment. Thus
Productivity – The features provided with voice based communications can allow users to communicate effectively and efficiently.
Open standards – With the help of open standards like SIP multi-vendor implementations are easily possible, opening up a window of choices for the enterprise.
The process of converting voice to packets
Now that we took a small history lesson into how we ended up with VoIP, let’s try to understand how a voice gets converted to a packet.
As you speak into a telephone our voice gets converted to analog in the form of electrical signals. Analog communication can be represented using a waveform like the one below.
The amplitude is measured on a scale of +127 to -127. These analog signals can be sampled to create equivalent values that represent the value of the amplitude at a particular time. How many samples need to be taken can be determined by a theorem postulated by Dr. Harry Nyquist. It was known as the sampling theorem and stated that to accurately represent an analog signal we have to sample at twice the highest frequency. Taking the highest frequency as 4000 Hz. We should take at 8000 samples per second (4000Hz. X 2). Each sample can be represented with a single byte. The first bit of the byte is used to represent the positive or negative scale and the next 7 bits (2^7 = 128) are used to represent the actual value.
Voice at this point has been digitized. If we calculate the bandwidth required we end up with 64 kbps (8 bits X 8000 samples/second). This conversion mechanism converts our voice into the G.711 PCM standard. There are various methods or mechanism in place to further reduce this bitrate. For example, G.729 codec consumes only an eight of the bandwidth (8 kbps). Various implementations of codecs exist each having their own advantages and disadvantages. One such example is the G.722 codec. Even though it uses the same bandwidth consumed by the G.711 codec (64 kbps) it produces noticeably better audio quality as it captures a wider range of frequencies.