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Glossary

TURN server

Also called: TURN relay, Traversal Using Relays around NAT

A TURN (Traversal Using Relays around NAT) server is a relay on the public internet that forwards media between devices when firewalls or network address translation stop them from connecting directly. It is the last-resort path in a WebRTC call.

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Why calls need a relay

WebRTC tries to send audio and video peer-to-peer. But most devices sit behind NAT, and some network setups make a direct path impossible: symmetric NATs that assign a new public port for every destination, carrier-grade NAT on mobile networks, and corporate or public Wi-Fi firewalls that block UDP or unusual ports. In those cases, both devices can still reach a server on the internet, so a TURN server sits in the middle and forwards packets between them.

STUN vs TURN

The two helper servers in WebRTC connectivity
Job
STUN:Tells a device its public IP and port
TURN:Relays all media between devices
Carries call audio/video?
STUN:No
TURN:Yes
Cost to run
STUN:Very low
TURN:High — every relayed byte crosses the server
Works when
STUN:NATs allow hole punching
TURN:Almost always, including strict firewalls

Both are IETF standards. TURN is an extension of STUN, and many servers do both.

How ICE decides to use TURN

  1. 1

    Gather candidates

    Each device collects possible addresses: host (its local address), server-reflexive (its public address, learned via STUN) and relay (an address allocated on the TURN server).

  2. 2

    Exchange them

    Candidates travel between the devices over the app's signaling channel.

  3. 3

    Check pairs

    ICE tests candidate pairs, generally preferring direct routes over relayed ones.

  4. 4

    Pick one

    The best working pair is used. If only relay pairs succeed, the call goes through TURN — over UDP, TCP, or TLS on port 443 when a firewall allows nothing else.

On Asyncbit

Asyncbit's voice and video calls use WebRTC with a TURN relay, so when a direct connection isn't possible — common on mobile data and restrictive networks — the call is relayed instead of failing to connect. Calls start from a chat, 1:1 or group, in the browser or on Android. Group calls hold up to 8 people, with video for up to 4; larger calls continue as audio. Chats get a summary line afterwards, such as “Missed audio call”.

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