OpenSIPS RTPEngine audio issue

Job ID: 35890818

Budget: $30 – $250 USD

Hi,

We have OpenSIP (with WSS) with RTPEngine configured but we are not able to make audio calls working for the webrtc based client.

Our flow of calls is like this:
WebRTC client -> OpenSIPS -> FreeSWITCH

The system is deployed on Azure.

We are looking for experienced person who has done such work and quickly help us.
Related categories: Asterisk PBX VoIP FreeSwitch