Full Rate

GSMGSM 06.10GSM Full RatelibgsmFRfull rate channel (FR)Full-Rate (FR)GSM-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

Global System for Mobile CommunicationsGSM networkGSM (850/900/1800/1900)
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

EFRGSM-EFREnhanced Full-Rate (EFR)
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 encodingspeech codecSpeech
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.

Ekiga

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

RTP payload formats

RTP audio video profileRTP audio video formatsRTP Audio Video Profiles

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 channel (HR)Half-Rate (HR)
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)

Linear predictive coding

LPClinear prediction coefficientsBlock Independent LPC
It uses linear predictive coding (LPC). Like many other linear predictive coding (LPC) speech codecs, linear prediction is used in the synthesis filter.

ETSI

European Telecommunications Standards InstituteEuropean Telecommunications Standards Institute (ETSI)Telecommunications in Europe
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 Predictionlong-term predictionregular pulse excited-long term prediction
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 linear predictive coding (LPC) 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

widewide-bandbroadband analog
In modern narrowband speech codecs the order is usually 10 and in wideband speech codecs the order is usually 16.

Pulse-code modulation

PCMLPCMLinear PCM
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

G711G.711.1ITU G.711
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-lawu-lawPCMU
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

PSTNpublic telephone networktelephone 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

Technische Universität BerlinBerlin Institute of TechnologyTU Berlin
"libgsm" was developed 1992–1994 by Jutta Degener and Carsten Bormann, then at Technische Universität Berlin.

Wine (software)

WineWine 1.0Wine software
This codec can also be compiled into Wine to provide GSM audio support.