G-Stream media platform has recently implemented support for WebRTC protocol.
We decided to tell in detail why this technology that is gaining popularity is necessary for those who want to deliver live broadcasts to anywhere in the world with minimal delay and no buffering.
The simplest way to use WebRTC in media streaming is by enabling sharing a stream without an application installed.
MPEG-DASH and HLS streaming implementations usually come with latency limitations. More often than not, you have to deal with a delay of a dozen seconds with these solutions. Since live and interactive are becoming more and more important in the modern-day world, broadcasters are looking to share their streams directly in the browser with no need to install any additional software.
RTMP audio and video streams are being sent to media platform, converted into WebRTC and distributed to end users via our CDN to enable low-latency streaming.
Connecting to a CDN with a live stream required Real-Time Messaging Protocol (RTMP) support, which translated into using an additional media gateway component. G‑Core Labs now provides such capabilities of connecting WebRTC to RTMP. Moreover, it enables encoding, transcoding, recording and scaling WebRTC-based streams which allows to reduce buffering and costs for broadcasters.
“Thanks to this technology and continuous improvement to the services provided by our company, we are making a breakthrough in the field of video broadcasting: users are getting closer to each other and recognize that everything is happening here and now”
To enable the option to broadcast content using WebRTC, contact your account manager or write to technical support via chat or email to email@example.com.