In the past, Voxbone was one of the industry pioneers of WebRTC connections as a complimentary service to SIP Trunks. We’re happy to have contributed to a technology we continue to believe in and, now that there’s an abundance of high-quality open-source WebRTC gateways and proxies available in the wild, we believe it’s best to take a step back and let the community develop in the direction it sees fit.
Even though our own WebRTC service was shut down in 2018, you can still access the building blocks of our WebRTC interface and all the necessary tools and tutorials needed to replicate the service on your own, at scale. Alternatively, try the fully supported LiveSwitch Cloud SDK available from the WebRTC experts at Frozen Mountain.
If you’re interested in setting this up and would like some advice, feel free to contact our product team or reach out to the good folks at Frozen Mountain for more information.
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:
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!