Production audio codecs
PCMU and PCMA are the production baseline for SIP trunks, recording and interoperability. Opus is supported for Webphone and internal WebRTC paths; a carrier trunk still requires its own acceptance test.
RFC 4733 is the supported DTMF method. SIP INFO is accepted only after a device or carrier-specific test.
- PCMU / PCMA
- Supported for negotiation and transcoding in the 1.0 FreeSWITCH image.
- Opus
- Supported for WebRTC and internal calling; transcode to the carrier's accepted baseline when required.
- G.729
- Not yet a production codec. Passthrough and BCG729 transcoding remain behind legal, image and load-test gates.
Supported browsers
RelayPBX supports the current and previous major release of Chrome and Edge on desktop, Firefox for panel and audio calling, and current/previous Safari on macOS and iOS. Current Chrome on Android is supported for the responsive panel and Webphone.
HTTPS, WSS, microphone permission and a working ICE/TURN path are mandatory. Internet Explorer, legacy EdgeHTML, embedded social-app browsers and unsupported operating systems are excluded.
Phones and provisioning
Template families exist for Yealink, Fanvil, Grandstream, Poly, Snom, Cisco MPP/SPA and Htek. A template means RelayPBX can render a secure bootstrap file; it does not by itself certify every physical model and firmware.
- Reference acceptance targets Yealink T33G/T46U/T54W, Fanvil X4U-V2/X5U-V2/X6U-V2 and Grandstream GRP2612P/GRP2614/HT812.
- Poly Edge E220/E350, Snom D735/D785, Cisco 8841/8851 MPP and Htek UC926U/UC924U remain template-available until physical acceptance is recorded.
- A certified result includes registration, RTP, DTMF, hold/MOH, blind and attended transfer, BYE, BLF, MWI and factory-reset provisioning evidence.