Re: [rtcweb] draft-sipdoc-rtcweb-open-wire-protocol-00 (Open In-The-Wire Protocol for RTC-Web)

[Roman Shpount <[e-mail]>]

On Sun, Oct 30, 2011 at 8:06 PM, Cullen Jennings <[e-mail]> wrote:

>
> On Oct 27, 2011, at 11:09 , Hadriel Kaplan wrote:
>
> > One process-based concern about making requirement 4 a WG requirement:
> you can't actually do SIP over Websocket with a "pure SIP network" until we
> get Websocket into a SIP-extending RFC as a new transport type.  I wouldn't
> want to hold up WebRTC docs becoming RFCs, waiting for the DISPATCH and
> probable SIPCORE process to make a websocket SIP transport into a RFC.  I
> *want* to add Websocket as a SIP transport type, but it's not actually as
> trivial as one would think.
>
>
I think a better idea would be to create a new type of server which works
as a proxy between websocket and arbitrary TCP/UDP type connection. This
way we don't need to break TURN servers to implement this. Proposing
something like this obviously outside of scope of WebRTC, and it also has
some interesting security issues (how do we proxy websocket to TLS socket
for SIPS?).
_____________
Roman Shpount