20 Apr
2016
20 Apr
'16
8:08 a.m.
Summary: I'm just getting started with VoIP/SIP/WebRTC. Following the FreeSwitch Cookbook I've built a webclient that uses SIP signalling to establish a connection with FreeSwitch and then exchanges media with WebRTC. But I get the impression from a lot of sources that SIP is a bad fit for WebRTC. One datapoint that indicates this is the very existance of mod_verto. Why would you bother writing it if SIP, that has 10+ years of battle testing, was a good fit right? What I don't understand yet is why SIP is a bad fit for WebRTC? Would someone in the know care to muse about that? regards Oivvio