Overview
Telephone calls have traditionally been routed through the Public Switched Telephone Network (PSTN), a technology for landlines dating back over a century. PSTN calls are limited to an 8kHz sample rate using a narrowband audio codec, resulting in audio that typically sounds muffled or lacks range.
Modern cell phones can use VoIP for calls when connected via Wi-Fi or mobile data. VoIP can leverage wideband audio codecs that transmit audio at a higher sample rate, resulting in much higher quality audio, often referred to as HD Voice.
LiveKit SIP supports wideband audio codecs such as G.722 out of the box, providing higher quality audio when used with HD Voice-capable SIP trunks or endpoints.
HD voice support
Support for HD voice is dependent on your SIP trunking provider. Currently, Telnyx is the only SIP provider that supports HD voice. To configure Telnyx for HD voice, see Configure HD voice.
LiveKit Phone Numbers supports HD voice out-of-the-box.