WebRTC One To Many Broadcasting (Audio & Video) on i.MX6 board
Browers like FireFox are providing WebRTC (Web Real-Time Communication) technology to do realtime audio and video communication. WebRTC can be used for peer to peer audio and video communication, one to many audio and video broadcasting. The WebRtc provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose. WebRTC offers web application provides rich, real-time multimedia applications (think video chat) on the web, without any plugins, downloads or installs.It’s purpose is to help build a strong RTC platform that works across multiple web browsers, across multiple platforms
iWave Systems has developed WebRTC based one to many audio and video Broadcast on i.Mx6 Q7 development platform.
iWave’s i.Mx6 Q7 platform has Quad core processor which can operate up to 1 GHz speed/core. i.MX6 CPU is freescale’s latest achievement in integrated multimedia application processors which is part of growing multimedia-focused products that offers high performance processing and are optimized for lowest power consumption.iWave’s i.Mx6 Q7 platform supports 1GB RAM in 64bit mode with eMMC memory of 4GB which can be used both as Mass storage and boot device. i.Mx6 Q7 also supports Ethernet port which is integrated i.Mx6 CPU and connected to the external Gigabit Ethernet PHY on SOM.
iWave’s Application consist of Server and clients similar to the Peer to Peer communication. In case of Broadcast there is only one way communication. The server side application should be run in Linux/Windows PC. The clients can access the server from Firefox browser using the following https://localhost:8888/index.html address. When a client wants to join a broadcast , the client will be registered first in the server . The video codec which is used in the WebRTC is VP8 with special video jitter buffer which prevents packet loss. Since VP8 is the codec used, it is well suited for RTC as it is designed for low latency. The audio codec used in WebRTC is OPUS codec. In this application user is given option to broadcast only audio or only video or both audio and video.
WebRTC Broadcast Call: A Screeshot of WebRTC one to many audio and video broadcast
- Mozilla Firefox Browser is available with built in WebRTC.
- Video and audio streaming quality through WebRTC is much improved.
- No Plugins or Packages need to installed along with Firefox to run WebRTC.
- The Streaming can be done with systems on local network.
- The processor used here is quad core because of which multiple task can be assigned at the same time.
A.Deepanraj - Member Technical
iWave Systems Technology Pvt. Ltd.