Configuring HD Voice for SIP

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 Opus and G.722 out of the box, providing higher quality audio when used with HD Voice-capable SIP trunks or endpoints.

Telnyx supports HD Voice for customers in the US. To enable HD Voice with Telnyx, ensure the following are configured in your Telnyx portal:

• HD Voice feature is enabled on the phone number you are trying to use (under Number -> Voice)
• Opus and G.722 codecs are enabled on your SIP Trunk (under SIP Connection -> Inbound)
  • We recommend leaving the other default codec enabled as well for compability

Currently, Twilio does not support HD Voice. If you find other providers that support HD Voice, please let us know so we can update this guide.