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
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