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TURN Settings

Web Real-Time Communication (WebRTC) provides high-quality video and audio communication capabilities in some web browsers, without requiring installation of a custom plug-in. By using Google Chrome, users both inside and outside your enterprise network can attend web-based Polycom ®  RealPresence ®  Web Suite conferences. In these conferences, media is exchanged directly between WebRTC clients (mesh conference) or between WebRTC clients and a Polycom MCU.

To support WebRTC-based video conferencing, the RealPresence DMA system implements both Session Traversal Utilities for NAT (STUN) and Traversal Using Relays around NAT (TURN) protocols. When needed, a RealPresence DMA edge-configured or combination-configured system can act as a STUN and TURN server to enable firewall and Network Address Translation (NAT) traversal of UDP media traffic between WebRTC clients.

TURN is necessary when a WebRTC client wants to communicate with a peer but cannot do so because both client and peer are behind respective NATs. STUN is not an option if one of the NATs is a symmetric NAT (known to be non-STUN compatible). TURN is also needed when direct UDP media cannot be exchanged for other reasons (for example, due to an organization's firewall policies). Using the TURN protocol, a WebRTC client can allocate a media relay port on the TURN server that the far end can use to indirectly send media to the WebRTC client.

When you enable and configure the TURN server and a TURN user, internal and external WebRTC clients can request TURN media relay services.

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