Private Cloud Streaming
Application that controls a variety of transport protocols in addition to video and audio codecs. Our Private streaming Incoming and outgoing live HD/4K streams can be routed and transcoded using the software user friendly interface. The client/server-based system, which can be expanded at any time and designed as needed, ensures maximum failure safety.
Individual authorisations can be defined in detail thanks to user and role management. Users can only view, edit, create, or remove the content that is shared with them thanks to this personalisation. The most crucial parameters, including bitrate, frame rate, and input and output streams, are shown at a glance in an inventive preview area.
Private Cloud Streaming I/O
- Transport streams over UDP, RTP, HLS, TCP, Webrtc and HTTP
- Multiple input and output streams
- AES encryption and decryption
- Configuration via web services/SOAP
- Software packages for server or cloud installation
| Redundancy switching of the input signal (Multicast) | ✔ |
| HLS streaming for iOS Android devices | ✔ |
| Crossover switch matrix | ✔ |
| Adjustable time shift and stream delay | ✔ |
| Support for multiple network interfaces | ✔ |
| IPTV live stream transcoding | ✔ |
| OTT internet broadcasting | ✔ |
| Transcoding for multi-screens and multi-devices | ✔ |
| RTMP | HLS | WebRTC | |
| Latency | 2-5 seconds | 20-30 seconds | <500 milliseconds |
| Scale | The system’s capacity is limited because of persistent server-client connections. Needs special RTMP proxy to scale | Millions of viewers | ≤10000 viewers |
| Quality | No ABR. Quality depends on the bandwidth availability of the clients | ABR enables excellent network adaptability and superior quality | With simulcast, quality can adapt to network conditions |
| Reach | Is nonexistent in the last mile delivery due to the decline and eventual death of Flash | It is compatible with all HTML5 players. Supported by all client platforms | Currently supported by most modern browsers, iOS, and the Android ecosystem |
RTMP
Adobe created the Dusk RTMP, also known as the Real-Time Messaging Protocol, to allow for high-performance live streaming of data, video, and audio between Adobe Flash Player and a specialized RTMP streaming media server. To put it simply, it can reliably transmit a video stream from the source to the viewer with an average delay of less than 10 seconds.
HLS / HTTP Streaming
HTTP Live Streaming, or HLS Any web browser or HTML5 player can play media delivered over HTTP using the HTTP Live Streaming Protocol, or HLS. All browsers, mobile devices, and LR platforms now use HLS extensively to make streaming and watching videos easier. One of HLS’s main advantages is its use of Adaptive Bitrate Streaming (ABR), which significantly improves stream quality and viewer experience. ABR enables each client to receive multiple versions of the same video. The client can select from various streams to create a seamless playback experience with improved quality and less buffering, depending on the network conditions. Stated differently, they can choose the quality of the video according to the strength of their network.
WebRTC
A free and open-source technology called WebRTC, or Web Real-Time Communication, makes it possible to communicate data, audio, and video in real time without the need for plugins. In short, it enables real-time audio and video communication within web pages. ABR is also partially supported. Depending on the network conditions of each client, WebRTC selectively forwards only one of the many video renditions that are produced.