SIP audio drops on BYOC when screen recording extension v2.1.4 is active

Voice & Telephony
1

Chrome flag #enable-features=WebRtcAllowH264HighProfile is enabled, but the extension v2.1.4’s still killing the SIP audio path with ERR_CONNECTION_REFUSED right at the orgs/{id}/media/recordings endpoint. Service worker logs dump codec negotiation failed on opus/48000 before the BYOC handshake ts, doing jack all for the call. Docs say v2.1.5 patches the manifest lock, yet the store’s stuck on 2.1.4.

2

Are you pulling the BYOC config from state or editing it inline?

Problem

The ERR_CONNECTION_REFUSED drops when the media handler forces opus before the handshake.

resource "genesyscloud_byoc_config" "sip" {
 media_handling = "passthrough"
}

Don’t fight the manifest lock, just switch to passthrough.