Hi everyone! We're doing some integrations with telephony systems and LiveKit. We're receiving audio in Opus format from telephony systems but while passing it to LiveKit, I see we're having to decoding it to raw PCM format, and while transferring the audio back, we're again receiving raw PCM from LiveKit and encoding to Opus to send telephony systems. The problem is encoding & decoding operations taking a lot of resources, is there any possibility to pass the audio in Opus format to LiveKit ? I see LiveKit uses WebRTC to transfer audio from client to its server, and in that WebRTC, I believe we negotiate the codec by specifying it in SDP, so far I believe LiveKit specifying raw PCM in negotiation SDP, so is there any possibility to use Opus in that negotiation and transfer audio in Opus over WebRTC ? If this is not available yet, we're ready to dedicate our time in contribution if any guidance provided, since it's being crucial for us. Also just curious question: why we're transferring audio in raw PCM over WebRTC ? wouldn't it be more efficient to transfer audio in compressed version like in Opus ? Thank you!

Can you use a different codec for the SIP side like G711?