Parameters#
The following VST configuration options can be modified in tokkio-app-params.yaml file. At least one valid turnserver (coturn/twilio) or valid reverse-proxy server should be provided for webcam and avatar streaming to work over WebRTC.
VST Configuration# stunurl_list
set list of stun URLs, It discovers their public IP and type of NAT
[“stun.l.google.com:19302”,”stun1.l.google.com:19302”]
static_turnurl_list
list of TURN servers with long term credential mechanism. TURN provides a relay mechanism for communication when direct peer-to-peer communication is not possible due to NAT traversal issues.
[“admin:admin@10.0.0.1:3478”, “admin:admin@10.0.0.1:3478”]
use_coturn_auth_secret
enable authentication secret mechanism for coturn
false
coturn_turnurl_list_with_secret
list of co-turn servers with short term credential mechanism. To use this config, use_coturn_auth_secret should be set to true
[“10.0.0.1:3478”:<secret_key>, “10.0.0.1:3478”:<secret_key>]
use_twilio_stun_turn
enable to use twilio stun and turn server
false
twilio_account_sid
twilio account username. The config option use_twilio_stun_turn should be set to true to enable use of twilio server
“”
twilio_auth_token
authentication token of twilio account
“”
use_reverse_proxy
use reverse proxy instead of turnserver (coturn/twilio). RP is public facing service that directly receives and handles client traffic, performing the appropriate routing so that the traffic arrives at tokkio cluster VPC.
false
reverse_proxy_server_address
if use_reverse_proxy is set to true, then set reverse_proxy server address & port. Also env variable REVERSE_PROXY_SERVER_ADDRESS can be used to set the ip_address.
10.0.0.1:100
max_webrtc_out_connections
set maximum count of webrtc out connections i.e avatar stream
8
max_webrtc_in_connections
set maximum count of webrtc in connections i.e webcam stream
3
grpc_server_port
set GRPC server port. GRPC server used to port negotiate with ov-renderer to receive avatar udp stream
50051
webrtc_in_audio_sender_max_bitrate
set max bitrate in kbps to be used by web-UI for webcam stream
128000
webrtc_in_video_degradation_preference
set degradation preference to be used by webrtc sender for webcam stream. It controls the quality of media streams (either resolution or framerate) when network conditions degrade
“framerate”
total_video_storage_size_MB
set max video record size used to record webcam stream
10000
always_recording
set always recording on or off
false
gpu_indices
set GPU indices to use particular gpu device inside vst container, If there are multiple gpu devices visible inside container
[]
webrtc_port_range
set webrtc min and max port range. This should be in sync with nodePort range in the VST microservice helm-chart. VST uses nodeports for webrtc media traffic due to separate RP instance
min 30001, max 30030
use_software_path
enable or disable software path, if gpu is not available.
false
enable_websocket_pingpong
enable websocket periodic ping pong.This is to avoid websocket connection break due to proxy/LB or any firewall policies
false
websocket_keep_alive_ms
websocket periodic ping pong time in milliseconds
5000