Codecs

Common VoIP Codecs
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 ofprocessor 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 | Robustto packet loss. Free |
Speex | 2.15 / 44 | Minimizes bandwidth usage by using variable bit rate. |

Codec Summary table | | |
Voice transmission is analogical, whereas the data network is digital. The process to sample analogical waves into digital information is made by an encoder-decoder (CODEC). There are many standards to sample an analogical voice signal into a digital one. Theprocess is often quite complex. Most of the conversions use pulse code modulation (PCM) or variations

In addition, the CODEC zip the sequence of data, and sometimes provides echo cancellation. The compression of the waveform can save bandwidth. This is especially interesting in low speed connections so you can have more VoIP connections at the same time. Another way to save bandwidth is using thesilence suppression. The goal is not to send packages when there is no voice in the conversations.
Next is a table with the most known codecs in use:

- Bit Rate - The rate at which bits are transmitted over a communication path. Normally expressed in Kilobits per second (Kbps)
- Sampling Rate - the number of samples taken per second when digitizing sound. The quality of the digital reproductionimproves as the number of samples taken per second increases.
- Frame size - The time between packets sent
- MOS - (Mean Opinion Score). It is a subjective measure of sound quality from 1 to 5.

In order to understand better the codec process and the parameters expressed in the table we recommended to read the section of G.711 codec process where it is possible to learned how it works theG.711 codec.

In order to understand better the codec process and the parameters expressed in the table we recommended to read the section of G.711 codec process where it is possible to learned how it works the G.711 codec.
Number | Standard by | Description | Bit rate (kb/s) | Sampling rate (kHz) | Frame size (ms) | Remarks | MOS (Mean Opinion Score) |
G.711 * | ITU-T | Pulse codemodulation (PCM) | 64 | 8 | Sampling | U-law (US, Japan) and A-law (Europe) companding | 4.1 |
G.721 | ITU-T | Adaptive differential pulse code modulation (ADPCM) | 32 | 8 | Sampling | Now described in G.726; obsolete. |   |
G.722 | ITU-T | 7 kHz audio-coding within 64 kbit/s | 64 | 16 | Sampling | Subband-codec that divides 16 kHz band into two subbands, each coded using ADPCM |   |
G.722.1 |ITU-T | Coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss | 24/32 | 16 | 20 |   |   |
G.723 | ITU-T | Extensions of Recommendation G.721 adaptive differential pulse code modulation to 24 and 40 kbit/s for digital circuit multiplication equipment application | 24/40 | 8 | Sampling | Superceded by G.726; obsolete. This is a completely different codec than G.723.1|   |
G.723.1 | ITU-T | Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s | 5.6/6.3 | 8 | 30 | Part of H.324 video conferencing. It encodes speech or other audio signals in frames using linear predictive analysis-by-synthesis coding. The excitation signal for the high rate coder is Multipulse Maximum Likelihood Quantization (MP-MLQ) and for the low rate...