The title of this bug seems to indicate it is limited to enabling TCP for the control connection, but to still use UDP for data relay. The ?transport=tcp parameter in a TURN uri is to signal request of TCP relay ports. Securing the control connection requires the uri scheme to be "turns" instead of "turn" . Specifying both should secure the transport as well.
+---------------------------------+----------+--------+-------------+ | URI | <secure> | <port> | <transport> | +---------------------------------+----------+--------+-------------+ | turn:example.org | false | | | | turns:example.org | true | | | | turn:example.org:8000 | false | 8000 | | | turn:example.org?transport=udp | false | | UDP | | turn:example.org?transport=tcp | false | | TCP | | turns:example.org?transport=tcp | true | | TLS | +---------------------------------+----------+--------+-------------+
Thanks
Oleg
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I'd like to clarify what "TCP/T:LS support" means for TURN:
RFC5766 defines only "one-sided" support for TURN protocol: the communications between the client and the TURN server may go over TCP/TLS, but the TURN server always relays the packets over UDP.
There is an extension to TURN, RFC 6062, that defines the "full" TCP support: the data "after" TURN can be relayed over TCP (not TLS). The connection between the client and the TURN server must be TCP or TLS if we want the traffic to be relayed over TCP.
Our new TURN server version supports this RFC 6062 now:
http://code.google.com/p/rfc5766-turn-server/
But I am not sure that this feature is useful for WebRTC traffic. I have no idea whether WebRTC is going to support "full" TCP relaying and whether it has a real use case for WebRTC.