I've been looking for documentation on how to build a WebRTC Gateway with Freeswitch (WebRTC to SIP/RTP) but I haven't found anything which could help me.

I found something on this maillist: https://lists.freeswitch.org/pipermail/freeswitch-users/2020-April/133228.html "FreeSwitch for WebRTC to SIP/RTP gateway?" so it seems it could be done but there is no documentation.

I'm trying to build a gateway which could translate webrtc into SIP in order to let some users connect their extensions to a PBX based on Asterisk that does not support WebRTC.

I'm looking for documentation but also a Company who could help us in the initial configuration.

Thank you very much.

Regards,

Manuel