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