OpenSIPS RTPEngine audio issue
Budget $30-250 USD
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.
4 freelancers are bidding on average $166 for this job
This might be an issue with webrtc connectivity with freeswitch SIP handle. Please go through my past freeswitch and VoIP projects and customer feedback over ten years
Hello, I have more than seven years of experience in the office and more than three years of freelance experience in the required task and would like to help you with this task. Thanks for posting in my area of work.