SIP registration pass to Genesys from Freeswitch

Hi Team,

I am using webrtc asterisk phone in my environment. For Webrtc ro work with Genesys, I need to register the extension in Genesys via Freeswitch. As of now we are using Audiocodes which transfers the webrtc SIP registration message to Genesys.

Now I need to the same activity in Freeswitch means I will send SIP registration message to Genesys via Freeswitch and I will not register that extension in FS instead it will get resistered to Genesys.

Then I guess you need Kamailio or OpenSIPS for that.

Actually OpeSIPS or Kamailio is just proxy, I need to convert WSS traffic to SIP so how can we use Freeswitch ?

Kamailio supports SIP over WSS too.

@dujinfang1 I want leads from sales team