Need help for WebRTC (sip.js) implementation

Job ID: 33727042

Budget: $30 – $250 USD

We are looking for an expert who can help us to make our webrtc client working with opensips. We have Opensips as SBC and FreeSWITCH to handle media + call routing logic.

If we connect our webrtc client with FreeSWITCH directly then webrtc working well but when we connect the webrtc client with opensips then outbound and inbound calls are not working.

Please bid only if you have worked on similar task.
Related categories: Asterisk PBX VoIP FreeSwitch SIP WebRTC