VT Receiver is a Windows app designed to receive feeds from VT Publisher, VT Guest and The Guest Link. VT Receiver can make feeds available as NDI® source, play out to a hardware device or create a virtual source.
Select one of the available sources in the "Source" tab (5).
Once selected, the preview of the feed will soon start to play.
Select desired protocol in tab 6 (SRT and H.265 are used by default).
Select desired latency in the "Buffer" tab (10).
Select the output method in the "Output" tab (NDI®, a hardware device, or a virtual device) and hit "Start Output" (15).
Here's what VT Receiver looks like.
The current version of the Video Transport software, the license count in your current license, its validity date and the latest release version.
The Publisher ID – your unique identifier, provides access to all feeds published under this ID. The Publisher ID is hardwired into your license and cannot be changed.
Copy and send to the remote contributor if you want someone to contribute via a browser. Learn more by reading about The Guest Link.
Connects to the submitted Publisher ID.
Lists the streams available (published) under this Publisher ID and license. Color indicator meaning: the stream is published and available (grey); the stream is being received currently (green); there is a problem with the stream (red).
Select the protocol and encoder to use for transmission: SRT-HEVC (default, requires an Nvidia hardware codec), SRT-AVC (H.264 encoding), and WebRTC (H.264 encoding).
Video parameters of the steam: video format, number of audio channels.
Current frame rate.
Buffer size to configure latency. Learn more about latency configuration.
Current latency. If less then 350 ms, shows two values: the network latency of the stream and real latency (network latency plus decoding latency).
Lost packets count and percentage.
Select the output destination: publish as NDI®, play out to a supported hardware device, or create a virtual device.
Set the output name of the stream.
Start and stop receiving selected streams.
Controls to enable full-screen preview on available monitors; mute control.
Decoding stats: current protocol (WebRTC or SRT), bitrate (Mbps), video codec, and FPS performance (value has to be higher than the stream's FPS).
There are several principal latency settings in Video Transport:
Real-time. In good conditions the technology is capable of bringing latency down to 50-100 ms. When using this option, make sure your network is good and stable – network issues will cause frame drops. Use Ultra-low if you want a compromise between reliability and latency on good connections.
Low. If reliability is more important than latency, increase the buffer. More latency gives our protocol more time for error correction (to resend the missing frames and restore the image). Use Reliable for predictable reliability even on low-quality connections.
Fixed. Depending on the quality of your network, you can choose one of the fixed buffer settings. This is the recommended mode for remote multi-camera production since it allows for several sources to arrive at the same time. Also to be used in creative scenarios, such as a group of musicians jamming together).