There are several ways that Video Transport is used by customers. Here's a good place to start learning about Video Transport use cases.
There's no one-fit-all answer, but you will find the basics in this article.
Generally, one channel is one active transmission, but there are some exceptions. Pleaser refer to this page for Video Transport licensing details.
To do this, you will need to configure mix-minus audio routing in the background (follow the link to learn now).
Yes, if the same fixed latency setting is used in the streaming parameters.
Yes. Please check the features page.
As of today, VT Publisher, VT Receiver, VT Server and VT Guest are available for Windows only.
But the Guest Link, the Web Preview and the Control Panel can be run in a browser on a Mac.
Not at this moment, but we are getting there. Stay tuned by signing up.
VT does not have a built-in comms features, although some users simply route the intercom via an extra audio channel, which can be selected on the receiving side (read more on audio routing).
However, we suggest to deploy a separate intercom system (ideally, running on separate hardware): for instance, if you lose the video feed and need to troubleshoot, you will also lose the intercom and won’t be able to do that. So it’s better for reliability reasons to separate video transport and intercom. One solution we suggest is Unity Intercom, but there are others available as well.
Even if you set 50 Mbps as your streaming bitrate, the encoder will use only the maximum bandwidth that is needed for this particular content. For instance, if the input is an upscaled SD stream or a white wall in 4K, a lot less bandwidth will be required to transmit this stream in it's original quality.
If you have made sure that you have the bandwidth you need on both locations, the reason may be that the connection to our TURN server is not good enough. If this is something you observe, please reach out and we will look for a solution. This is valid only for WebRTC connections.
There are no limitations for this in the app itself, but the performance of your PC and GPU card are usually limited. Please check our post on system requirements.
Yes, and there are two reasons:
The apps have access to the H.265 codec (see technical requirements for hardware details). The browser-based implementation is limited to H.264 and VP9.
The apps default to the SRT protocol with tons of improvements done by our team. In the browser we are limited to WebRTC.
The benefit of the Guest Link is that it works on all platforms including mobile devices.
This option is not available in any of the Video Transport apps, but it is easy to do using Direct Take – our multi-channel video recording solution.
You can use Dante Virtual Soundcard in WDM mode and use Dante audio as the External audio for video sources. In WDM mode Dante Virtual Soundcard appears in the system as the number of separate stereo devices, so there is a limit of having only 2 channels of External audio per video stream, you can't select more then one External audio device per video stream.