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Configure Freeswitch to have high quality audio Opus codec through WebRTC to be like Jitsi

Hi Fellow VoIP experts,

We are working on a leading web-conferencing open source platform for online learning.

This platform uses Freeswitch with Opus codec through WebRTC; so directly through the browser.

We made some tests and the audio quality is a lot better on Jitsi, we want to achieve the same quality through our platform.

If you are a certified VoIP engineer or have a lot of experience configuring Freeswitch for web usage with opus codec through WebRTC we would be glad if you could help us.

So your mission is to help us understand Freeswitch mod_conf opus codec and achieve a better audio quality result.

Skills: VoIP, Asterisk PBX, Linux, FreeSwitch

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About the Employer:
( 2 reviews ) Mourouj 3, Tunisia

Project ID: #27371607

2 freelancers are bidding on average $225 for this job

sparxsols11

Good day! Hello! I am very happy to get this opportunity to bid for your project.i have gone through your project description. "This is my area of specialization, i can get it done perfectly well. So drop me a line ov More

$250 USD in 7 days
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langemecdev

Hi. I am video conference developer and developed myslef directly LMS site. also, with Asterisk and Freeswitch, I have rich experience. My MLS site uses BigBlueButton and BigBlueButton More

$200 USD in 7 days
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