On Wed, May 6, 2015 at 11:01 AM, Brett Tate <br...@broadsoft.com> wrote:
> Concerning RFC 7118, "wss" was not defined as transport for SIP-URI. If > dialog setup with "transport=ws" as a SIP-URI parameter within Contact or > Record-Route entry when the connection is actually secured using wss (such > as within RFC 7118 section 8.2 example), should wss continue to be used? > Based upon RFC 7118 section 8.2 example (such as ACK sent over wss), I > assume yes; however I didn't notice anything within RFC 7118 discussing > the topic. > > Does this matter if SIP Outbound (RFC 5626) and GRUU (RFC 5627) are used? SIP over WebSockets is useless without SIP outbound or some other protocol that causes mid-dialog requests to flow over the existing connection established by the client. Regardless if connection is WS or WSS, the same connection will have to be used by the server, since it cannot establish a new connection to the client. In real implementations the value of the Contact is only used to populate the mid-dialog request URI or request URI for new calls directed to the client. It is not used to determine the message destination or the actual transport being used. I would also think that the Contact URL in that example should have been SIPS, not SIP if WSS was used. And that the subsequent forwarding should have been over TLS, not UDP. P.S. I think there is a great need of some sort of informational document that explains how SIP is supposed to work in NAT scenarios with connection oriented protocols, such as TCP. TLS. or WebSockets. It is a very confusing and typically mis-implemented scenario. _____________ Roman Shpount _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
