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# helpdesk
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cc @great-umbrella-58130 @echoing-kitchen-90064
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Heya Serhii, this is mostly correct, except for the fact there is no redirect between video servers. If a user is routed to video1.my.com by the load balancer, but the room they're requesting has already been created on video2.my.com the user will stay connected to video1.my.com for signaling tasks but the rtc connection by the users' client will be directly to video2.my.com. Here's a similar thread with some more info: https://livekit-users.slack.com/archives/C025KM0S1CK/p1677776934967869?thread_ts=1677746380.612349&cid=C025KM0S1CK
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g
to add to dan’s point, you don’t need a DNS address for each of your LiveKit Server instances. a single DNS address for the load balancer will suffice for signaling/wss. and for RTC, clients will connect directly to the nodes via public IP address.
l
Thank. You. But what about this config section ? Does it has to be identical on every livekit instance ? turn: # Uses TLS. Requires cert and key pem files by either: # - using turn.secretName if deploying with our helm chart, or # - setting LIVEKIT_TURN_CERT and LIVEKIT_TURN_KEY env vars with file locations, or # - using cert_file and key_file below # defaults to false enabled: true # defaults to 3478 - recommended to 443 if not running HTTP3/QUIC server # only 53/80/443 are allowed if less than 1024 udp_port: 3478 # defaults to 5349 - if not using a load balancer, this must be set to 443 tls_port: 5349 # needs to match tls cert domain domain: livekit.example.com # optional cert_file: /opt/livekit.deploy/certs/cert.crt key_file: /opt/livekit.deploy/certs/key.key
g
TURN is optional, and a separate from what we’ve been talking about above. see https://docs.livekit.io/oss/deployment/#improving-connectivity-with-turn
l
@great-umbrella-58130 turn we need in case if participants are in different networks and cannot directly connect to each other. In SFU case we are in different networks with livekit server. For example I deployed it on my PC and want to connect to a call from my phone using cellular network. Right ? Just there are few things which still confuse me in webrtc 😅 . Maybe somebody has good articles with explanations for this subject
g
TURN is more for restrictive firewall rules, say you want to connect to LiveKit Server behind a restrictive corporate firewall
if you’re deploying LiveKit Server on your PC and want to connect from your cell phone, you would just need to make sure the ports are open for both signal/wss and media/rtc