What is Opus codec bitrate for WebRTC call?

For one WebRTC call we have 4 opus streams:
1) Chrome to WCS
2) WCS to VoIP
3) VoIP to WCS
4) WCS to Chrome

Each stream is 20ms opus 48kHz coded.
Each opus packet has length about 100-150 bytes.
So, avarage bitrate is 100-150 * 1000/20ms * 8 = 40-60kbps
According Wireshark, avarage bitrate is 52 kbps.

It seems it’s 48-64 kbit/s for FB mono music.
Please, see section 2.1.1 of opus RFC for more information: http://tools.ietf.org/pdf/rfc6716.pdf