Voxbone launches WebRTC Beta Program


In the past, Voxbone was one of the pioneers in the industry to offer WebRTC connections as an alternative to SIP Trunks. We’re happy to have contributed to the technology we believed in, and now that there’s an abundance of open-source WebRTC gateways and proxies out there, we believe we’ve done our part.

However, even though our WebRTC service was shut down in 2018, you can still get all the building blocks of our WebRTC interface and all the necessary tools and tutorials needed to replicate the service on your own, at scale.

If you’re interested in setting this up and would like some advice, feel free to contact our product team.

Original Article:

Ten lucky Voxbone customers will test out WebRTC as a front-end for customer service or conferencing and run the traffic over the high quality Voxbone network. WebRTC is an exciting Open Source initiative spearheaded by Google to bring Real-Time Communications to every Web browser and IP enabled device. We are thrilled about WebRTC and want to be a part of it!

Initially, Voxbone plans to enhance WebRTC by delivering calls originated on the Web to SIP networks via our high quality network. Over the past week we had the privilege of briefing many of the industries most savvy analysts, bloggers and journalists about the Voxbone’s WebRTC news.

Here is a sample:

No Jitter

Light Reading

We especially appreciated Sarah Reedy’s own comment:

“It works! I must say, it is rare for demos to go smoothly the first time, but Voxbone’s did. It was literally just click this link and you’re on the call. That kind of ease of use will be important to make WebRTC viable. Excited to see more companies taking advantage of it. Makes thinges easier on desktops and mobiles.”

Thanks so much for the positive and encouraging feedback!