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VoIP Codecs Explained - How to Improve Call Audio Quality
https://telnyx.com/resources/codecs-affect-voip-sound-quality
G.711 VoIP codecs. The G.711 codec offers some of the best sound quality. However, it also requires the most bandwidth. This codec requires at least 96Kbps of bandwidth per line. If you want slightly better audio quality, you can use this codec with less compression.
What Are VoIP Codecs & How They Affect Call Sound …
https://www.nextiva.com/blog/voip-codecs.html
A VoIP codec is a technology that determines the audio quality, bandwidth, and compression of Voice over Internet Protocol (VoIP) phone calls. VoIP codecs use either proprietary or open-source algorithms. The word codec is a portmanteau of two terms: Co mpression and Dec ompression.
Silk Audio Codec to Transmit Speech Over Internet
https://www.vocal.com/speech-coders/silk/
SILK is designed by Skype as an internet wideband audio codec for use in VoIP. It operates at four different sampling rates: 8 kHz narrowband, 12 kHz mediumband, 16 kHz wideband, and 24 kHz superwideband. These allow for the capture of higher frequencies, which provide fuller sound, while also allowing interoperability with the Public Switched ...
EETimes - Error-Resilient Coding for Audio …
https://www.eetimes.com/error-resilient-coding-for-audio-communication-part-3-fec-techniques-for-speech-and-other-coding-techniques/
Indeed, while the ITU and the GSM have standardized several CELP codecs to use at different rates, in telecommunication systems some of the primary VoIP systems use waveform-based codecs. For example, Microsoft Messenger uses …
FreeSWITCH And The Opus Audio Codec mod opus
https://freeswitch.org/confluence/download/attachments/12517398/Freeswitch_and_the_Opus_Audio_Codec_1.0.pdf?version=1&modificationDate=1477328153855&api=v2
Besides being a high quality and low latency audio codec, the main features of Opus for VOIP are FEC and the ability to negotiate asymmetry of audio streams. The mod_opus module supports two variants of Opus: one at 48 KHz which can be used for WebRTC and it's present in the default configuration, and one which enforces the sampling rate at
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