STUN and TURN Solution


iconScalable STUN and TURN Service

STUN and TURN are the required components of any WebRTC communications solution. STUN is used for negotiating a peer-to-peer connection and TURN kicks in when the peer-to-peer connection is not possible, for example in cases where one or both of the parties is behind a firewall. Around 15% of all voice and video traffic is relayed through TURN servers. In such cases, close geographic proximity to a TURN server and low latency connectivity become increasingly important.

  • Reliable, scalable STUN
  • Price-friendly, geo-distributed TURN
  • Supports standards RFCs 3489, 5389, 5769, 5780, 5766, 6062, 6156, 5245, 5768, 6336, 6544, 5928 over UDP, TCP

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WebRTC STUN and TURN capabilities

Lowest Latency

With TURN nodes deployed with 4 providers in 14 different regions around the world, the lowest possible latency is guaranteed.

Reliable

Our servers are distributed over many data centers, utilizing multiple failover and load balancing strategies.

Price-friendly

Enjoy the forever free STUN and 1 GB free monthly TURN for playing around and testing. Scale as you grow, while our best available prices go down with the volume.

Simple integration

It takes a single API call to get your secure TURN credentials. Simply use them in your existing code when creating a WebRTC PeerConnection!

Contact us to discuss your use case !

  • STUN & TURN
  • Voice & Video
  • IP Messaging
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