Full Rate

GSMGSM 06.10GSM Full RatelibgsmFRfull rate channel (FR)Full-Rate (FR)
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system.wikipedia
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GSM

GSM (850/900/1800/1900) GSM/GPRS/EDGEGSM technology
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system.
It may also refer to the (initially) most common voice codec used, Full Rate.

Enhanced full rate

EFREnhanced Full-Rate (EFR)ETSI GSM enhanced full rate
Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate.
Enhanced Full Rate or EFR or GSM-EFR or GSM 06.60 is a speech coding standard that was developed in order to improve the quite poor quality of GSM-Full Rate (FR) codec.

Adaptive Multi-Rate audio codec

AMRAMR-NBAdaptive Multi-Rate
Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate.
There are a total of 14 modes of the AMR codec, eight are available in a full rate channel (FR) and six on a half rate channel (HR).

Speech coding

speech codecspeech encodingspeech
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system.
Full Rate, Half Rate, EFR, AMR for GSM networks

Ekiga

The GSM 06.10 is also used in VoIP software, for example in Ekiga, QuteCom, Linphone, Asterisk (PBX), Ventrilo and others.
Audio codec algorithms: iLBC, GSM 06.10, MS-GSM, G.711 A-law, G.711 µ-law, G.726, G.721, Speex, G.722, CELT (also G.723.1, G.728, G.729, GSM 06.10, GSM-AMR, G.722.2 [GSM‑AMR-WB] using Intel IPP)

Ventrilo

Ventrillo
The GSM 06.10 is also used in VoIP software, for example in Ekiga, QuteCom, Linphone, Asterisk (PBX), Ventrilo and others.
Ventrilo supports GSM Full Rate and Speex codecs.

Half Rate

GSM-HRHalf-Rate (HR)HR
Half Rate
Since the codec, operating at 5.6 kbit/s, requires half the bandwidth of the Full Rate codec, network capacity for voice traffic is doubled, at the expense of audio quality.

Linphone

The GSM 06.10 is also used in VoIP software, for example in Ekiga, QuteCom, Linphone, Asterisk (PBX), Ventrilo and others.
Audio codec support: Speex (narrow band and wideband), G.711 (µ-law, A-law), GSM, Opus, and iLBC (through an optional plugin)

ETSI

European Telecommunications Standards InstituteEuropean Telecommunications Standards Institute (ETSI)ETS
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.

Long-term prediction (communications)

long-term predictionregular pulse excited-long term predictionRPE-LTP
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.

Linear prediction

linearlySignal predictioncoefficient
Like many other speech codecs, linear prediction is used in the synthesis filter.

Narrowband

narrow-bandnarrownarrow band
In modern narrowband speech codecs the order is usually 10 and in wideband speech codecs the order is usually 16.

Wideband

widebroadband analogsurface/wideband
In modern narrowband speech codecs the order is usually 10 and in wideband speech codecs the order is usually 16.

Pulse-code modulation

PCMlinear PCMLPCM
The speech encoder accepts 13 bit linear PCM at an 8 kHz sample rate.

Analog-to-digital converter

ADCanalog to digital converteranalog-to-digital conversion
This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table.

G.711

ITU G.711A-lawG.711.1
This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table.

A-law algorithm

A-lawPCMA
This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table.

Μ-law algorithm

μ-lawµ-lawu-law
This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table.

Public switched telephone network

PSTNtelephone networkpublic telephone network
This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table.

Technical University of Berlin

TU BerlinTechnische Universität BerlinTechnical University
"libgsm" was developed 1992–1994 by Jutta Degener and Carsten Bormann, then at Technische Universität Berlin.

Wine (software)

WineCorelWineMS-DOS
This codec can also be compiled into Wine to provide GSM audio support.

Winamp

Bento BrowserWinamp 5.58Winamp3
There is also a Winamp plugin for raw GSM 06.10 based on the libgsm.