VideoRoaming Distributed STUN/TURN Service

Enhance the global quality and reach of your SIP and/or WebRTC service with the VideoRoaming Distributed STUN/TURN service featuring 11 Access Points, global redundant Quality of Service network, and optimized Internet transit.

Embed Global Reach and Quality in Your SIP or WebRTC Service

To embed the VideoRoaming distributed STUN/TURN in your video client 
(SIP or WebRTC), codec or service, it is recommended the standards listed below are followed and that a slight adjustment to ICE behavior is implemented.


VideoRoaming provides a globally distributed STUN/TURN service. The service is accessed using IP anycast, ensuring an optimal media path from all parts of the world.

The STUN/TURN servers are compliant with the following IETF RFCs:

RFC 5389 Session Traversal Utilities for NAT (STUN)
RFC 5766 Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)
RFC 5780 NAT Behavior Discovery Using Session Traversal Utilities for NAT (STUN)
RFC 6156 Traversal Using Relays around NAT (TURN) Extension for IPv6

ICE Considerations

In addition to providing clients with NAT traversal assistance, VideoRoaming utilizes TURN as a mechanism to attract media into the VideoRoaming Quality of Service (QoS) network.

In order to take full advantage of this service, and ensure media is routed via the VideoRoaming QoS network when placing calls outside the user's local network, a slight adjustment to standard ICE behavior is needed. By simply disabling the server-reflexive ICE candidate one make sure that media is routed via the VideoRoaming TURN servers whenever a call is placed outside the user's local network.


WebRTC is not yet fully compliant with the latest TURN IETF RFC (RFC 5766) with respect to the distributed nature of the VideoRoaming STUN/TURN service. A short-term fix is to hard code the server address to the closest TURN server.

The TURN credentials currently need to be hardcoded in the WebRTC JavaScript making it accessible to anyone. We plan for a fix enabling time-limited secret-based authentication for WebRTC applications.



APIs are available for creation/management of TURN users and extraction of TURN traffic records.

Developer Credentials

Complete the form below to get free developer credentials and start testing/embedding the VideoRoaming distributed STUN/TURN service:

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