GL Announces Comprehensive Codec Support for Telecom Networks
Gaithersburg, Maryland, USA – June 28, 2016 - GL Communications Inc., announced today supported Voiceband Codecs for its TDM, VoIP and Wireless Products.
Speaking to press, Mr. Vijay Kulkarni, the CEO of the company said, “The word "codec" is a compression of the two words "coder decoder" and is used to identify a voice compression algorithm. Engineers are always looking to transmit or store voice in the least bandwidth and time - thus the need for more sophisticated codecs. Applications include: Wireless networks where efficient use of scarce frequency spectrum is important, VoIP networks where packetized and compressed voice is used to transmit over the World Wide Web, PCs, MP3 players, and mobile phones where codecs are used to store and play music.”
He added, “GL Communications’ products support a variety of signaling and audio processing applications. Using these tools, one can emulate, analyze, and troubleshoot audio signaling over TDM, IP and Wireless Platforms. These tools support a wide variety of Voiceband Codecs including the narrow-band, and wideband (HD audio) codec standards.”
Mr. Kulkarni further explained, “GL’s TDM platform products support codecs like µ-Law, A-law (G.711) and these are the versions of PCM (Pulse Code Modulation) have been the standard throughout the world for digital voice transmission for telephony since early 1960’s. Voice is filtered between 300 to 3400 Hz, then sampled at 8000 samples/second, with 12 to 13 bits per sample, then companded to 8 bits/sample resulting in 64 kbps.
G.726 codec, the Adaptive Differential Pulse Code Modulation (ADPCM) is originally, a half-rate alternative to ITU-T G.711 and includes both the G.721 and G.723 standards. G.726 compresses by converting between linear, A-law (used in Europe) or µ-Law (used in the U.S and Japan) PCM and 40, 32, 24 or 16 kbps.
G.722 is a wideband speech codec standard operating at 48, 56 and 64 kbps with an encoding frame length of 10 ms. It samples audio data at a rate of 16 kHz (using 14 bits) with an encoding frame length of 10 ms, double that of traditional telephony interfaces, that provides superior audio quality with clarity.
He added, “GL’s IP platform products support a wide variety of codecs and to discuss some them, GSM-FR is a Full Rate speech coder standardized by the ETSI to compressing toll quality speech (8000 samples / second) and it was the first digital speech coding standard used in GSM digital mobile phone systems.
GSM-EFR (6.60) is an extended version of GSM-FR (6.10) codec. With sampling frequency of 8000 samples/sec and frame size of 31 bytes, it achieves the bit rate of 12.2 kbps with an encoding fixed frame length of 20 ms. GSM Half Rate (HR) 6.20 operates with sampling frequency of 8000 samples/sec. It outputs the frames of size 14 Bytes, that puts the bit rate of encoder at 5.6kbps with an encoding frame length of 20 ms.
Adaptive Multi-Rate Speech Codec (AMR) is the 3GPP standard codec for narrowband speech and multimedia messaging services over GSM and evolved GSM (WCDMA, GPRS and EDGE) networks. It operates at eight bit rates in the range of 4.75 to 12.2 kbps with an encoding frame length of 20 ms and it was specifically designed to improve link robustness.
AMR-WB provides improved speech quality because of a wider speech bandwidth that is of 50–7000 Hz compared to narrowband speech coders which in general are optimized for POTS wireline quality of 300–3400 Hz.
Selectable Mode Vocoder (SMV) codec compresses each 20 milliseconds of 8000 Hz, 16-bit sampled speech input into output frames of one of the four different sizes: Rate 1 (171 bits), Rate 1/2 (80 bits), Rate 1/4 (40 bits), or Rate 1/8 (16 bits) with an encoding frame length of 20 ms. SMV is the preferred speech codec standard for CDMA2000, and also used in third generation handsets.
internet Low Bit Rate Codec (iLBC) codec is suitable for real time communications such as, telephony and video conferencing, streaming audio, archival and messaging. The codec supports two basic frame lengths: 30 ms at 13.33 kbit/s and 20 ms at the rate 15.2 kbit/s, using block independent linear-predictive coding (LPC) algorithm.
SPEEX Narrow Band (NB) is based on CELP Narrowband (8 kHz with an encoding fixed frame length of 20 ms) open source codec specifically used for VoIP and file-based applications.
SPEEX Wide Band (WB) has a sampling rate of16000 samples/sec with an encoding fixed frame length of 20 ms, which makes it a wide band codec.
E-Model based Mean Opinion Scores (MOS) and R-Factor score are two commonly used methods to measure the quality of voice over IP network are also supported by the aforementioned codecs.”
Mr. Kulkarni further added, “GL’s wireless platform products support a whole range of codecs and most of them are common to our TDM and IP product range as well like AMR (NB, WB), EVRC, EVRC-B, ECRC-C, GSM (HR, FR) and more. Recently introduced Enhanced Voice Services Codec (EVS) codec provides vastly improved voice quality, network capacity and advanced features for voice services over LTE and other radio access technologies standardized by 3GPP. It is the first 3GPP conversational codec providing up to 20 kHz audio bandwidth, offering speech quality that of highest standard.
EVS codec includes a multi-rate audio codec, a source controlled variable bit-rate (SC-VBR) scheme, a VAD, a comfort noise generation (CNG) system, and an error concealment (EC) mechanism to offset the effects of transmission errors resulting in lost packets. Its channel-aware mode feature further improves frame/packet error resilience.”