Full Rate

GSM-FR is specified in ETSI 06.10 (ETS 300 961) and is based on RPE-LTP (Regular Pulse Excitation - Long Term Prediction) speech coding paradigm.

The codec operates on 160 sample frames that span 20 ms, so this is the minimum transcoder delay possible even with infinitely fast CPUs and zero network latency.

[1] The free libgsm codec can encode and decode GSM Full Rate audio.

[2][3] "libgsm" was developed 1992–1994 by Jutta Degener and Carsten Bormann, then at Technische Universität Berlin.

[5][6] The GSM 06.10 is also used in VoIP software, for example in Ekiga, QuteCom, Linphone, Asterisk (PBX), Ventrilo and others.