Before the age of VoIP, analog voice signals in traditional telephones travelled electronically from the sending telephone line to the receiving line in a closed-path circuit that remained open throughout a call connection. Sounds patterns such as voice are converted into electromagnetic form through a technique called analog modulation in the transmitter circuitry of the telephone device, and the analog signals are converted back into audible sounds by the receiving telephone device at the terminating end.

We later learned to convert the analog signals to digital form – or basically into ones and zeros that represent data in the computing world – to transmit sound over long distances. This paved the way for many telephone and computer advances that ushered in circuit-switched networks and packet-switched networks that eventually led to VoIP technology.

Harry Nyquist – a pioneering scientist who introduced sampling theorem that is considered the foundation of digital networking – later developed a device called coder-decoder (codec) that basically does the work of converting sound and video into digital form and vice versa. Note that codecs work differently from modulator-demodulator (modem) devices, as the latter takes digital signals from computers and converts these to analog signals for transmission over analog telephone lines.

VoIP codecs

Today’s codecs come as software that does the same conversion of signals into digital form and back. There are codecs for handling audio, video, fax and text transmissions, and those most commonly used with VoIP are the following:

G.711

Also called pulse code modulation (PCM), this codec has a high bit-rate at 64kbit/s and is considered the language of modern digital telephones in such standard platforms such as H.320 and H.323. The G.711 codec renders accurate speech signals and requires low processor power. This codec is useful in VoIP systems as a method of fax transmission, though it requires a high 128 kbps bandwidth for voice calls.

G.722

This codec can be used with varying compressions (48/56/64 kbit/s) to conserve bandwidth and adapt to changing network congestion. It provides superb audio clarity with its 16kbps sampling rate – double that of regular telephone interfaces – and is useful for VoIP in LANs where bandwidth is readily available.

G.723.1

Mostly used in VoIP systems for its low bandwidth requirement (5.3/6.3 kbit/s), this codec offers high compression and high-quality audio. It can be used in a dial-up environment, though it also needs lots of computer processing power.

G.726

This is an improved version of G.721 and G.723 codecs, which adapts to multiple bandwidth (16/24/32/40 kbit/s) and is used in international telephone trunks and in some wireless phones for bandwidth conservation.

G.729

This codec is error-tolerant and offers superb bandwidth utilization at 8kbit/s. This VoIP-capable codec requires license for use in devices and costly in terms of the processing time it needs to convert signals.

GSM

This free codec is the same encoding used in GSM cellphones and is available in many hardware and software platforms. It has a high compression ratio and low bandwidth requirement at 13kbit/s.

iLBC

Stands for Internet Low Bitrate Codec, this is a royalty-free (under a fairly liberal license) speech codec with 15kbit/s bandwidth, with a robust voice communications capability.

Speex

This is an open-source audio codec that is designed primarily for speech. With an awesome range of 2.15 kbit/s to 44 kbit/s, this flexible VoIP codec minimizes bandwidth usage but requires CPU processing power.

Saiju George is an IP communications and hosted business software entrepreneur. He’s always on the lookout for great business services like a free SIP proxy service for free IP calls, or a great mobile SIP solution.