OpenSIPS RTPEngine audio issue

Job Description:


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.

Навыки: VoIP, Asterisk PBX, FreeSwitch

О клиенте:
( 4 отзыв(-а, -ов) ) Ahmedabad, India

ID проекта: #35890818

4 фрилансеров(-а) готовы выполнить эту работу в среднем за $166


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

$250 USD за 1 день
(53 отзывов(-а))

I appreciate the Job Employment Invitation. I understand your requirement of Open SIPS RTP Engine audio issue. About VSOnline Services: We are a custom software development firm with 7+ years of extensive hands-on exp Больше

$135 USD за 7 дней(-я)
(9 отзывов(-а))

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.

$140 USD за 999 дней(-я)
(0 отзывов(-а))