Re: WebRTC has landed
On 24/02/13 00:20, Timo Juhani Lindfors wrote:
> Daniel Pocock <daniel@pocock.com.au> writes:
>> JavaScript and give users of their package the ability to click'n'call
>> other users within the web page.
>
> Have you had time to study how the technology works? If both parties are
> behind a HTTP proxy (for example at an airport or cafe with WIFI) I
> think you need some third party server to relay the traffic? Are there
> any existing implementations that do this with webrtc?
>
>
It is based on two things:
- WebSockets carries the SIP signaling (e.g. to register the user
location, find the person you want to call). WebSockets works through
HTTP proxies
- ICE/STUN/TURN is used to find a way to get RTP streams working between
the peers. If you are both in the same airport, ICE can figure that out
and make your devices talk to each other without any relay. If one
person is in an obscure location, the TURN server will act as a relay
Both of these mechanisms require some kind of public server of course:
WebSockets requires a SIP proxy on a public IP address with a WebSockets
port open. The HTTP proxy in the airport/hotel, etc, doesn't know you
are talking to a SIP proxy though, because it looks like a HTTPS session.
ICE requires a TURN server (which needs two public IP addresses).
However, both types of server are relatively easy to set up if you have
a VPS somewhere, you can find a (Debian biased) guide on
http://www.RTCquickstart.org
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