In our configuration, we use the Asterisk confbridge with SIP and Opus codec. Clients connect to Janus over WebRTC and then Janus uses SIP to connect to the Asterisk. All Janus is doing is forwarding RTP packets between the client and Asterisk.
We are noticing that if you induce packet loss in the RTP packets to Asterisk (even a small amount), the audio quality significantly deteriorates compared to when clients connect directly to each other over WebRTC (same packet loss) and use an SDP with same properties. The SDP offer sent to Asterisk and answer received both have useinbandfec=1. When there is RTP packet loss to Asterisk, it sounds like a crackling sound for the packets lost. It’s almost like the PLC is not working in Asterisk or it’s not using the FEC sent by the clients. Note: there is no packet loss for any packets sent from Asterisk (only 1 way so we can control this test). This is a link to a short sound recording of how it sounds: asterisk_sound_packet_loss.mp3 - Google Drive
Below are the configurations we use. Does anyone have any idea on why there is this audio quality degradation in Asterisk? It seems like there is some misconfiguration.
max_playback_rate=16000 ; Limit the maximum playback rate on the encoder