Properties reference
VT Publisher manager properties (Global settings)
Name | Format / Value example | Description | Publisher Global Settings equivalent |
---|---|---|---|
vt20:webrtc | true / false | New WebRTC implementation for the Web Preview and Web Guests. On by default. | VT 2.0 WebRTC mode |
vt20:ice_port_range | integer | Allows you to set custom port range for ICE (p2p) connection between Publisher and Receiver | VT 2.0 SRT/WebRTC port range (min-max) |
vt20:connect.gateways_filter | "+xxx.xxx.xxx.xxx:7777" | Use this prop to alter the VT SRT Gateways list. You can use ip address and port 7777 | VT 2.0 gateways filtering (e.g. for custom routes) |
runtime:gpu_pipeline | true / false | Enable gpu_pipeline mode if you using the BMD cards and 10bit feature. In other cases, it is not recommended to use it. | Turn on gpu_pipline (required for 10bit feature) |
device:bmd.10bit | true / false | Set this prop to TRUE to capture 10 bit video. This feature is only supported on Blackmagic Design devices. | Capture 10bit video (BMD cards only) |
device:bmd.444 | true / false | Set this prop to TRUE to capture 4:4:4 video. This feature is only supported on Blackmagic Design devices. | Capture 4:4:4 video (BMD cards only) |
asio.split_audio_channels | integer with commas | Split one ASIO audio device to many by the mask. Mask is applied for each device simultaneously. Format: X,X,X,.... Where X is the number of channels assigned from real device to split. For example, you have a device with the channels:0,1,2,3,4,5,6,7 - 7 in total. So if you set the Split audio channels=3,2,3 VT will create 3 audio devices with channels from the original device - 0,1,2; 3,4; 5,6,7 | Split audio channels (ASIO devices only) |
webrtc:venc_srt_props:output.444 | true / false | Enable this prop to encode the 4:4:4 video. | Encode 4:4:4 video |
webrtc:venc_srt_props:encoder.quicksync | true / false | If you enable this prop VT Publisher will use the QS265/264 encoder if your NVIDIA GPU reached the limit for encoding sessions. It should work for H265/H264 encoders. Works only with Intel CPU with an integrated HD graphics. Currently in beta. | Use Quicksync encoder |
vt20:connect.timeout | integer | Sets the timeout for the connection to signaling server. By default it is set to 5000 ms. | Set the timeout for connection (in milliseconds) |
screen_capture:capture.audio_device | microphone / PC Sounds (Loopback) | Selects audio source to use with a screen capture feed | Audio source for screen capture |
screen_capture:capture.frame_size | 1280x720 / 1920x1080 / … | Sets the resolution of screen capture feed | Screen capture resolution |
screen_capture:capture.fps_limit | example: 30.0, 60.0 | Sets the screen capture frame rate in frames per second | Screen capture frame rate |
screen_capture:capture.show_mouse | show / hide | Selects show or hide cursor on the screen capture video | Show cursor on the screen capture video |
screen_capture:capture.highlight_mouse | show / hide | If set to show , highlights mouse cursor with yellow circle | Show mouse highlight on the screen capture video |
screen_capture:capture.show_clicks | true / false | If true it will show animation if mouse clicked | Animate clicks on screen capture video |
preview:preview.type | dx9 / dx11 | Here you can select render API for the native in-app Preview Window | Set the preview type |
preview:overlay_rms | true / false | Enable this to add RMS audio meters overlay on the preview to observe the sound levels. | Show RMS audio meters on preview |
webrtc:venc_srt_props:preset | high_performance / high_quality | Sets the preset for NVIDIA encoder: High Quality or High Perfomance | Set the preset for Nvidia encoder |
vt20:connect.ice | true / false | Enable or disable P2P connectivity (ICE) | VT 2.0 P2P Connect Enabled |
audio.list_input_sources | true / false | If TRUE shows the list of local audio input devices in the VT Publisher sources list along with video devices | Show local audio input devices |
VT Publisher channel properties (Source settings)
Name | Format / Value example | Description | Publisher Source Settings equivalent |
---|---|---|---|
channel_label | textbox | Sets the channels name, requires republish. | Channel Label (require republish) |
webrtc:vt_tunnel.default | example: tcp://ip_address:port | Sets the address for UDP/TCP tunneling feature. | UDP/TCP Tunnel address |
webrtc:venc_srt_props:codec | GPU_H265 / GPU_H264 / CPU_H264 | Here you can select the desirable encoder for your stream from: GPU_H265, GPU_H264, CPU_H264. If you have a proper NVIDIA GPU we recommend you to use GPU_H265. In case if you don't have an NVIDIA GPU, better option is to use CPU encoding, but it will requires powerfull CPU. | SRT Default Encoder |
|
| Not working, will be deprecated. | Create Virtual Device for video source |
format.input | example: YUY2 640x480@25.00p 4:3 / … | Sets source video format. Auto by default. If Auto not working set this manually from the list. | Source Video Format |
external_audio | example: PC Sounds (Loopback) | Select external audio device | External Audio |
vt20:connect.public_port | textbox | Here you can manually set the ICE connection port for VT Publisher | SRT Publisher Port |
webrtc:audio_bitrate | example: 320K | Sets the audio bitrate for the source feed (per each 2 channels) | SRT Audio Bitrate per stereo-pair |
audio_channels_mask | example: 1,2 | Here you can set the audio channels which you wish to publish with the source. For example: 1,2,5,6 means that you selected channels 1,2,5,6 and only those channels will be heard in the stream. | Audio Channels Mask |
audio_gain | integer | Applies audio gain to source audio tracks. | Audio Gain |
web_preview:mix_minus.enabled | true / false | Automatic mix-minus for the Web Guests. | Web Guest Mix-Minus enabled |
web_preview:mix_minus.src_channels | integer | Sets the channels that Web Guests will hear in mix-minus | Web Guest Mix-Minus channels mask |
web_preview:mix_minus.delay_msec | integer | Delay for Mix-Minus. Default is 50 msec. You can raise up this value in case of crackling audio problems.Web Guest Mix Minus Delay - it's web guest audio delay time which is going to be used during the mix munus processing, by default 50 ms, so all remote peers will hearing another remote peers audio with that delay | Web Guest Mix-Minus delay msec |
web_preview:webrtc.external_decoder.h264 | true / false | Use the GPU decoding on Web Guest side if available. Improves perfomance and quality of the incoming stream on the Web Guests side. | Web Guest GPU H.264 decoder |
web_preview:video_encoder | VP9 / CPU_H264 / GPU_H264 | Select the encoder from the drop down list. GPU H264 is recommended option when you have an powerful NVIDIA GPU, because it encodes one time for all viewers. Other options are: VP9, CPU H264. | Web Preview & Guest video encoder |
web_preview:video_bitrate | example: 1.5M / 5.0M / … | Sets the bitrate for the Web Preview&Guest encoder in Megabits per second | Web Preview & Guest Max output feed bitrate |
web_preview:max_width | example: 1920 | Sets feed maximum width resolution for the Web apps | Web Preview & Guest output feed max width |
web_preview:max_height | example: 1080 | Sets feed maximum height resolution for the Web apps | Web Preview & Guest output feed max height |
video | true / false | Audio Only feed | Output Video Enabled |
audio | true / false | Video Only feed | Output Audio Enabled |
vt20:connect.gateways_filter | "+xxx.xxx.xxx.xxx:7777" | Use this prop to alter the VT SRT Gateways list. You can use ip address and port 7777. In Receiver this field should be empty by default. | VT 2.0 gateways filtering (e.g. for custom routes) |
vt20:ice_port_range | integer – integer | Allows you to set custom port range for ICE (p2p) connection between VT Publisher and Receiver | VT 2.0 SRT/WebRTC port range (min-max) |
channel_name.show_location | true / false | Adds the VT Publisher's location name to the source name in the list. | Show location in channel name |
runtime:gpu_pipeline | true / false | Enable this switch if you want to receive a video feed in 10bit. Disabled by default. | Turn on gpu_pipline (required for 10bit feature) |
webrtc:login_timeout | integer | Sets the maximum amount of time needed to establish a connection between your VT Receiver and a signaling server. Default value is "3000" (milliseconds). | Set the timeout for connection (in milliseconds) |
webrtc:decoder.output.10bit | true / false | Enables the 10bit video output feature. Requires gpu_pipeline setting. False by default. | Allow to output 10bit video |
preview:preview.type | dx9 / dx11 | Here you can select render API for the native in-app Preview Window. Options are dx11, dx9 and dshow. Default value is is "dx11" | Set the preview type |
preview:overlay_rms | true / false | Enable this to add RMS audio meters overlay on the preview to observe the sound levels. "False" by default. | Add RMS audio meters on preview |
vt20:connect.ice | true / false | Allows you to enable or disable P2P connection | VT 2.0 P2P Connect Enabled |
asio.split_audio_channels | Integer , integer … | Split one ASIO audio device to many by the mask. Mask is applied for each device simultaneously. Format: X,X,X,.... Where X is the number of channels assigned from real device to split. For example, you have a device with the channels:0,1,2,3,4,5,6,7 - 7 in total. So if you set the Split audio channels=3,2,3 VT will create 3 audio devices with channels from the original device - 0,1,2; 3,4; 5,6,7 | Split audio channels (ASIO devices only) |
access_id | string | Value contains the Access_ID for the stream. You can GET it. But it generated automatically. So if you try to SET it when you restart the stream it will be reset to new automatically generated value. | |
webpreview_permanent_id | string | Value contains the Web PreviewURL for the stream. You can GET it. But it generated automatically. So if you try to SET it when you restart the stream it will be reset to new automatically generated value. |
VT Publisher security settings properties
Name | Format / Value example | Description | Security Settings equivalent |
---|---|---|---|
auth::enabled | true / false | Allow anonymous access. False – Anonymous access True – Only users from pre-created user list are allowed | Allow anonymous access |
active | true / false | If False - user account is not active and not alllowed to see and participate in this feed, True - user account is active and allowed to see and participate. | Active |
guest_password | string | Sets access password for selected user account | Password |
con_limit | integer | Number of maximum connections from this account | Connections Limit |
start_time | datetime format yyyy/mm/dd hh-mm-ss | Time from when account is allowed to access the feed. | Start Time |
exp_time | datetime format yyyy/mm/dd hh-mm-ss | Time from when account can't access the feed. | Expiration Time |
To create the Security Settings User account, set Channel Props:
Name:
auth::
(for exampleauth::guest_from_sample
)Value:
guest_password='password' con_limit='number' start_time='yyyy/mm/dd hh:mm:ss' exp_time='yyyy/mm/dd hh:mm:ss' active='true / false'
(for exampleguest_password='123' con_limit='1' start_time='2022/05/16 18:31:00' exp_time='2022/05/16 21:31:00' active='true'
)
VT Receiver manager properties (Global settings)
Name | Format / Value example | Description | Receiver Global Settings equivalent |
---|---|---|---|
vt20:connect.gateways_filter | "+xxx.xxx.xxx.xxx:7777" | Use this prop to alter the VT SRT Gateways list. You can use ip address and port 7777. In Receiver this field should be empty by default. | VT 2.0 gateways filtering (e.g. for custom routes) |
vt20:ice_port_range | integer – integer | Allows you to set custom port range for ICE (p2p) connection between VT Publisher and Receiver | VT 2.0 SRT/WebRTC port range (min-max) |
channel_name.show_location | true / false | Adds the VT Publisher's location name to the source name in the list. | Show location in channel name |
runtime:gpu_pipeline | true / false | Enable this switch if you want to receive a video feed in 10bit. Disabled by default. | Turn on gpu_pipline (required for 10bit feature) |
webrtc:login_timeout | integer | Sets the maximum amount of time needed to establish a connection between your VT Receiver and a signaling server. Default value is "3000" (milliseconds). | Set the timeout for connection (in milliseconds) |
webrtc:decoder.output.10bit | true / false | Enables the 10bit video output feature. Requires gpu_pipeline setting. False by default. | Allow to output 10bit video |
preview:preview.type | dx9 / dx11 | Here you can select render API for the native in-app Preview Window. Options are dx11, dx9 and dshow. Default value is is "dx11" | Set the preview type |
preview:overlay_rms | true / false | Enable this to add RMS audio meters overlay on the preview to observe the sound levels. "False" by default. | Add RMS audio meters on preview |
vt20:connect.ice | true / false | Allows you to enable or disable P2P connection | VT 2.0 P2P Connect Enabled |
asio.split_audio_channels | Integer , integer … | Split one ASIO audio device to many by the mask. Mask is applied for each device simultaneously. Format: X,X,X,.... Where X is the number of channels assigned from real device to split. For example, you have a device with the channels:0,1,2,3,4,5,6,7 - 7 in total. So if you set the Split audio channels=3,2,3 VT will create 3 audio devices with channels from the original device - 0,1,2; 3,4; 5,6,7 | Split audio channels (ASIO devices only) |
VT Receiver channel properties (Source settings)
Name | Format / Value example | Description | Receiver Source Settings equivalent |
---|---|---|---|
extra_audio_output | example: Default Audio Device(WASAPI) | Allows you to send audio from stream to audio device | Extra Audio Output |
webrtc:vt_tunnel.default | example: tcp://ip_address:port | Sets the address for UDP/TCP tunneling feature. | UDP/TCP Tunnel address |
webrtc:decoder.nvidia | true / false | Allows you to use NVIDIA GPU for decoding the receiving feeds. "true" by default. | Use Nvidia Decoder |
webrtc:decoder.quicksync | true / false | Allows you to use Intel Quicksync decoder for decoding the receiving feeds, Disabled by default. | Use Quicksync Decoder |
video | true / false | Enables video output Enabled by default. | Output Video Enabled |
audio | true / false | Enables audio output. Enabled by default. | Output Audio Enabled |
vt20:connect.webrtc_transport | true / false | Allows you to use WebRTC to connect to stream in VT Receiver | WebRTC transport |
format.convert | true / false / non-standard | Convert the output into a selected format. Disabled by default. | Convert output format |
scaling_quality | auto / gpu | Output format conversion quality | Output format conversion quality |
format.output | Auto, PAL, HD720-50p, HD720-59p, HD720-60p, HD1080-25p, HD1080-29p, HD1080-30p, HD1080-50i, HD1080-59i, HD1080-60i, HD1080-50p, HD1080-59p, HD1080-60p | Select the desired video format to convert on output. Note that this parameter will only apply when the 'Convert output format' prop is "true" | Output Video Format |
audio_channels_mask | example: 1,2 | Enter the Audio Channels numbers that you want to output. | Audio Channels |
audio_gain | integer | Applies custom gain to the audio tracks | Audio Gain |
vt20:connect.public_port | integer | Specify the custom port number that you previously set on the Publisher side of this channel. Empty by default. | SRT Receiver port |
rcv_buffer | integer – integer | Here you can set the custom buffer value. | SRT Receiver Buffer (min-max) in msec |
device::out_name | string | Allows you to set the output name for the stream(works with NDI output case) or get the output name for other devices like SDI. | Stream output name. See the VT Receiveer manual. |
Web Guest Remote settings properties
Name | Format / Value example | Description | Web Guest Settings equivalent |
---|---|---|---|
name | string | Web Guest name | Name |
location | string | Web Guest Location | Location |
resolution | 320x240 / 640x360 / 640x480 / 1280x720 / 1920x1080 | Web Guest camera resolution | Resolution |
frameRate | 15 / 30 / 60 | Web Guest camera framerate | Frame Rate |
audioInput | example: 6d9995c58016bc6fad3c07feae2b2977fbd17071e1d7fdb03227ba982c56a937 | Web Guest audio input device | Select a mic |
videoInput | 22b0e5d30f57ff3054b90638b0015fecaacff1ef9961ee628043e8d36718658a | Web Guest video input device (camera) | Select a camera |
videoBitrate | 0.5M / 0.75M / 1.0M / 1.5M / 2.5M / 5M / 7.5M / 10M / 12.5M / 15M / 20M | Sets video Bitrate | Video Bitrate |
encoder | VP9 / H264 | Web Guest feed encoder | Video Encoder |
audioBitrate | 6k / 10k / 20k / 40k / 96k / 192k / 510k | Web Guest feed audio bitrate | Audio Bitrate |
mirroring | true / false | Enable Web Guest camera mirroring | Mirroring |
audioChannels | 0 / 1 / 2 / 3… | Audiochannels selection. Important! Value - is a stereo pairs count number, which starts from 0 ( i.e. 0 = channels 1,2, 1 = channels 2,3 etc.) | Audio Channels |
cameraMuted | true / false | Web Guest camera switch | Camera Off |
micOff | true / false | Web Guest mic switch | Microphone Off |
audioEnhancements | true / false | Enable or disable WebRTC AudioEnchancements | Audio Enhancements |
otherChannels | "1,2" \ "3,4" | Other channels selection | Other channels |
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