A codec is a coder-decoder software which converts an audio signal into compressed digital form for transmission and back into uncompressed audio signal for replay. Codecs (Audio & Video) play an important role in VoIP. Audio signals are sampled several thousand times per second by codecs before conversion and transmission over the network. These samples are again reassembled for producing the uncompressed audio signal. This way audio and video signals are transmitted over VoIP networks.
It is important to note that there are different sampling rates in VoIP namely 64,000 times per second, 32,000 times per second, and 8,000 times per second. So, different codecs use different sampling rates for conversion. For instance, G.711 codec uses 64,000 times per second sampling rate for conversion whereas G.729A codec uses 8,000 times per second sampling rate for conversion.
Codecs use advance algorithms to help sample, sort, compress
and packetize audio data. CS-ACELP algorithm is one of the prominent algorithms
used in VoIP. Basically, it organizes and streamlines the available bandwidth. The
algorithm intelligently defines whether to send data or not. This is precisely
why packet switching is superior to circuit switching. However, in VoIP it’s
not just codecs but softswitch plays an equally important role as it exactly
routes the call to intended destinations based on the IP addresses. So, retail
VoIP providers should always ensure that the codecs processed through their
endpoints are always compatible with that of the softswitch for guaranteeing
compelling VoIP calling experience to end users.
Some of the well-known codecs in VoIP are given below.
Codec
|
Bandwidth/kbps
|
Comments
|
G.711
|
64
|
Delivers precise speech
transmission. Very low processor requirements. Needs at least 128 kbps for
two-way.
|
G.722
|
48/56/64
|
Adapts to varying compressions and
bandwidth is conserved with network congestion.
|
G.723.1
|
5.3/6.3
|
High compression with high quality
audio. Can use with dial-up. Lot of processor power.
|
G.726
|
16/24/32/40
|
An improved version of G.721 and
G.723 (different from G.723.1)
|
G.729
|
8
|
Excellent bandwidth utilization.
Error tolerant. License required.
|
GSM
|
13
|
High compression ratio. Free and
available in many hardware and software platforms. Same encoding is used in
GSM cellphones (improved versions are often used nowadays).
|
iLBC
|
15
|
Robust to packet loss. Free
|
Speex
|
2.15 / 44
|
Minimizes bandwidth usage by using
variable bit rate
|
With this knowledge, a retail VoIP provider can make better
selection of softswitch as well as endpoints for their VoIP business.