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VoIP 101

I originally posted this at http://www.318.com/TechJournal

As the name implies, VoIP refers to voice or phone calls that traverse data networks using Internet Protocol (IP). This may mean that the calls are going over the Internet, or it may simply mean calls are traveling over privately managed data networks that are using IP to transport the calls from one location to the other.

This represents a fundamental change or shift in transportation and routing of traditional voice services work over analog wires.

With VoIP, the voice stream is broken down into data packets, compressed and sent to its destination using the Internet (as opposed to establishing a ‘permanent’ connection for the duration of the call), with routes traffic use depending on the most efficient paths given network congestion. Once received the packets are reassembled, decompressed, and converted back into a voice stream.

Digital format can be better controlled: we can compress it, route it, convert it to a better format, and so on. Digital signal is more noise tolerant than analog, but is also effected more by environment than Analog. VoIP applications require real-time errorless data streaming to support an interactive data voice exchange. This is obtained using Quality of Service (QoS). QoS helps ensure that packets aren’t lost, resulting in the loss of segments of voice traffic, or annoying clicks to users.

The bandwidth overhead for VoIP is far less than that of standard streaming audio. Today, every sound card allows 16 bit conversion from a band of 22050 Hz (for sampling you need a freq of 44100 Hz according to the Nyquist Principle) obtaining a throughput of 2 bytes * 44100 (samples per second) = 88200 Bytes/s, 176.4 kBytes/s for a stereo stream. Therefore, very good quality streaming audio requires 176.4 kBytes/s of bandwidth.

For VoIP, the throughput to send voice packets (176kBytes/s) . Digital data can be converted to a standard format that can be quickly transmitted, such as Pulse Code Modulation.

Pulse Code Modulation (PCM) is known to the IEEE as Standard ITU-T G.711 Voice bandwidth is measured at 4 kHz, so sampling bandwidth has to be 8 kHz (for Nyquist). Each sample is at 8 bits. Bandwidth requirements are 8000 Hz *8 bit or 64 kbit/s, as a typical digital phone line. Because of lower overhead and easier control over traffic, VoIP is cheaper in terms of bandwidth than using standard phone lines.

When using a standard phone line (PSTN), users pay a line manager company for the time used. The more time they talk, the more they pay. With VoIP Services, users can talk as long as they want with multiple people within their same VoIP network. For example, if a company has an office in Lansing, Michigan and an office in Los Angeles, California and use VoIP with their phone services then all calls between the Lansing and Los Angeles offices should be free.

Telephone companies use VoIP for a lot of long distance connectivity. They setup lines between two cities and are then able to transmit all calls between those two city’s free of charge with a much lower overhead than with PSTN lines. Vonage has entered the VoIP market, targeting residential services. By having sites in many of the major cities they are able to transmit calls between those cities free of charge.

Many companies also have sites in multiple cities as well. The availability of VoIP is now in a stable and mature stage and readily available from multiple vendors. Pricing has come down drastically over the past few years and VoIP solutions are now available for Small and Medium Sized business as well as the Enterprise.