# Audio Codecs *** ## OPUS *** The OPUS codec is supported when streaming in RTSP(S), SRT, and Multicast UDP protocols. If you will be viewing the stream in a web browser via WebRTC, use this codec for the best quality. ## AAC *** AAC is supported in all streaming protocols. If you will be viewing the stream in a web browser make sure to view it via HLS. HLS supports AAC audio while WebRTC does not. ## G711 *** The G711 codec is locked to a sample rate of 8000hz and mono audio. Both RTSP(S) and RTMP(S) support G711 while SRT does not. ## Compatibility Matrix *** | | OPUS | AAC | G711 | |---------------|------|-----|------| | RTSP(S) | Yes | Yes | Yes | | RTMP(S) | No | Yes | Yes | | SRT | Yes | Yes | No | | Multicast UDP | Yes | Yes | No |