OpenSIPS RTPEngine audio issue

Closed Posted 1 year ago Paid on delivery

$30-250 USD

Paid on delivery

Closed Paid on delivery

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.

VoIP Asterisk PBX FreeSwitch

Project ID: #35890818

About the project

4 proposals Remote project Active 1 year ago

Looking to make some money?

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4 freelancers are bidding on average $166 for this job

amelantoney's Profile Picture
amelantoney

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 in 1 day
(53 Reviews)
5.5
stylesiva's Profile Picture
stylesiva

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 More

$135 USD in 7 days
(9 Reviews)
4.6
SyedRohaanAlam's Profile Picture
SyedRohaanAlam

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 in 999 days
(0 Reviews)
0.0
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