I’m trying to setup an eavesdrop call between connected PSTN calls and WebRTC clients, and the audio doesn’t transcode.
The scenarios I need are:
- WebRTC (opus@48k) eavesdropping a PSTN call (PCMA@8k)
- PSTN (PCMA@8l) eavesdropping a WebRTC (opus@48k) call
The first example might be workable by forcing the WebRTC call down to 8k, but we don’t want to degrade the quality on an ongoing WebRTC call to listen via PSTN.
Am I missing something?