G.723.1 is used for compressing speech or other audio at a very low bit rate over public telephone (POTS) networks. It produces digital voice compression levels of 20:1 and 24:1 (6.3 Kbps MP-MLQ and 5.3 Kbps ACELP).

G.711 is a PCM scheme that works at an 8 kHz sample rate with 8 bits per sample, which requires 64Kbps data bandwidth. It can encode frequencies between 0 and 4 kHz and and is commonly used to encode the 4kHz analogue signal that defines "toll-quality" speech. ^[37]

G.726 is an ADPCM codec that provides 4Kbps speech bandwidth and switchable to 40, 32, 24 or 16kbps data streams. Its overall compression ratio of 2.8:1, 3.5:1, 4.67:1, or 7:1 for 14-bit linear data at 8 kHz and 1.6:1, 2:1, 2.67:1, or 4:1 for 8-bit compounded data at 8 kHz.^[45]

G.722 uses a SB-ADPCM codecs and supports 50 to 7kHz audio at 16 kHz sample rate with 64 Kbps bandwidth. It supplies narrow bandwidth audio either up to 7.5k in mono over a single 64 Kbps B channel or up to 15k in mono over two B channels.^[40] The two ADPCM sub bands used by this standard give audio performance superior to a single band ADPCM algorithm operating at the same bit rate. ^[43] The overall compression ratio of the G.722 audio coder is 4 to 1. ^[40]

G.728 using LD-CELP method to supplies 16 Kbps algorithm for telephone-bandwidth speech codec. It provides speech quality that is equivalent to or better than that of the G.726 32 Kbps ADPCM standard. G.728 is designed to meet the needs of low-delay high-quality speech coding and the overall delay is less than 2 ms. So it can be used in the environment of multiple speakers and background noise, and, also, to handle non-speech signals. ^[49]

MPEG-1 supports mono, stereo and dual mono sounds at 32, 44.1, and 48 kHz sampling rate. The predefined bit rate is from 32 to 448 Kbps for Layer I, from 32 to 384 Kbps for Layer II and from 32 to 320 Kbps for Layer III.

MPEG-2 support up to 5 main channels and a 'low frequent enhancement' channel and is backwards compatible to MPEG-1. Its bit rate is extended up to 1 Mbps; Furthermore, It also is an extension of MPEG-1 for lower sampling rates 16, 22.05, and 24 kHz for bit rates from 32 to 256 Kbps (Layer I) and from 8 to 160 Kbps (Layer II & Layer III).

MPEG-2 AAC provides a very high-quality audio coding standard for 1 to 48 channels at sampling rates of 8 to 96 kHz, with multi-channel, multilingual, and multi-program capabilities. AAC works at bit rates from 8 Kbps for a monophonic speech signal up to in excess of 160 Kbps/channel for very-high-quality coding that permits multiple encode/decode cycles.^[81]

G.723.1, G.726, G.722 output small bit rate and is suitable be used in the low band width network such as POTS. They are also recommended in LANs. G.711, G.722 are suitable to be used in ISDN and LAN. Whereas MPEG1, MPEG2 and MPEG2 AAC, with high sample rate and flexible out bit rate are reasonable to be used in LANs. Table 2-7 lists these audio codecs and their characters.

Table 2-7 Comparison of different audio codec methods