On 2014-12-16 17:29, James Knott wrote:
On 12/16/2014 11:21 AM, James Knott wrote:
Incidentally, it's not just browsers. It's an open protocol supported by W3C and IETF and can be implemented in other applications. So, you could have a video phone app on your smart phone that uses WebRTC. It's a means of getting away from proprietary protocols, such as Skype, or relying on a server, where the NSA etc., can get their paws on your conversations. Even if they intercepted your call somewhere on the Internet, they'd still have to break the encryption that's part of the spec. Compare that with regular SIP voice over IP calls, where encryption is generally not used.
And SIP is also peer to peer. The signaling goes via a server, at least initially, to find one another. Then the conversation can go directly end to end, no intermediary, or indirectly, via a host server; asterisk does this, but not for traversing firewall and nat, because it is done also intranet; it is done as a codec conversion service, so that both sides, even when using different codecs, can talk (I'm thinking of hardware voip phones which can not easily get new codecs). Firewall/nat traversal is done with the help from stun servers. http://en.wikipedia.org/wiki/STUN That direct conversation happens is obvious when you setup the whole thing yourself, and the server simply does not have the internet pipe to hold all the bandwidth of the simultaneous conversations it handles. Being a private setup, you control it fully. -- Cheers / Saludos, Carlos E. R. (from 13.1 x86_64 "Bottle" at Telcontar)