On-Demand WebRTC Tunneling in Restricted Networks

Abstract

In this paper we present the implementation of a WebRTC gateway service that can forward ad-hoc RTP data plane traffic from a browser on one local network to a browser on another local network. The advantage compared to the existing IETF STUN (RFC 5389), TURN (RFC 5766) and ICE (RFC 5245) protocols is that it does not require a public host and port mapping for each participating local host, and it works with more restrictive firewall policies. WebRTC implements ICE which combines STUN and TURN probes to automatically find the best connection between two peers who want to communicate. In corporate networks, simple hole punching and NAT traversal techniques typically do not work, e.g. because of symmetric NATs. Dynamic allocation of ports on an external 3rd party relay service is also typically blocked on restricted hosts. In our use case, doctors at hospitals can only access port 80 through the hospital firewall on external machines, and they need to communicate with patients who are typically behind a NAT in a local WiFi network. VPN solutions only work for staff but not between patients and staff. Our solution solves this problem by redirecting all WebRTC traffic through a gateway service on the local network that has a secure tunnel established with a public gateway. The public gateway redirects traffic from multiple concurrent streams securely between local gateway services that connect to it. The local gateways also communicate with browsers on their local network to mimic a direct browser-to-browser connection without having to change the browser runtime. We have demonstrated that this technique works well within the hospital network and arbitrary patient networks, without the need for any individual host configuration. In our evaluation we show that the latency overhead is 18-20 ms for each concurrent stream added to the same gateway service.