RadioKit Electron (beta)

About RadioKit Electron (beta) Android App

Electron takes the latest achievements in Audio over IP, to bring you ultimate tool for broadcasting, so you leave behind all of yours expensive hardware codecs.

Finally the revolution is done - you can now really exchange your outside broadcasting setup for a good smartphone, tablet, or a laptop and manage all of your transmissions through a web-browser.

And forget about Shoutcast/Icecast connotations, as this technology has nothing to do with it.

Sounds to beautiful to be true? Ask us for access and test it yoursefl - [email protected]

For all the geeks & nerds, here is tasty techtalk:

- The near-to-zero latency combined with website control allows to take
live broadcasting to a new era.

- Built upon Erlang – the language for building cellural networks.
Using Real Time Protocol and Real Time Streaming Protocol over UDP
– no more unreliable TCP connections and shoutcast streams.

- User interface based on React.JS – technology created by facebook,
made for the best comfort of snappy, collaborational communication.

- No hardware needed – everything is a software. Moreover: software
that's easy to integrate thanks to open web & mobile-ready APIs.

- Relies on well-tested cloud technologies and infrastructure: Amazon,
Azure & Digital Ocean clouds & the biggest European datacenter: OVH.

- State-of-the art Opus 1.1 codec that brings better quality than HE-AAC+1
with ultra low-latency and packet loss concaelment.

Supported operating systems & architectures:

Android 4.1 or newer
Windows XP or newer
Mac OS X 10.10 or newer, 64-bit (coming soon)
Ubuntu Linux 14.04 LTS, 64-bit (coming soon)

Networking:

Client-server architecture
No servers required by default (servers are hosted by RadioKit in the cloud)
No network setup required by default
Built-in NAT traversal (UDP hole punching)
VPN access (coming soon on-demand)
IPv6 ready
Designed to operate with cable, WiFi and mobile (3G and newer) IP networks

Codecs:

Opus 1.1 (RFC 6716)
Packet Loss Concealment enabled
Error Correction enabled
Bitrate from 5 to 510 kbit/s
- speech has broadcastable quality at 24 kbit/s
- music has broadcastable quality at 48 kbit/s
Sample rate from 8 to 48 KHz
Audio bandwidth: from 4 (narrowband) to 20 kHz (fullband)
Server-side audio processing engine internally uses 32-bit float samples at 48 KHz
Mono or stereo
Dynamically adjustable bitrate, audio bandwidth, and frame size

Transport layer:

Open standards: RTP (RFC 3550), RTSP 1.0 (RFC 2326)
Transmission happening over UDP whenever possible with TCP fallback

Latency:

Default latency: 80 ms + network latency:
- Codec frame size: 20 ms
- Receiver jitter buffer: 20 ms
- Server-side audio processing engine: 20ms
- Receiver transmission buffer: 20ms
- Network latency (60 ms on average in Central Europe)
Can be adjusted in the following range:
- Codec frame size: 2.5-60 ms
- Receiver jitter buffer: 5-1000 ms
- Receiver transmission buffer: 5-1000 ms
- Network latency: down to +/- 20 ms

Routing:

Multiple clients
Bi-directional audio communication with clients
Talkback

Authentication:

Based on OAuth2 standard
Built-in web-based authentication system

Monitoring & management:

100% web-based monitoring panel
Support multiple users operating collaboratively
Web-based peakmeter showing reference signal from the server
Ability to adjust connection parameters via web browser during transmission

Extras:

Built-in alternative communication channel: text chat Integration
All functions of the application are available through REST or WebSocket API
Open specification

Interested? - contact us on [email protected]