In the last few months I have been observing how the course for WebRTC is turning out so far . Unfortunately contrary to my expectations the fundamental holes in WebRTC specification are still the same with less being done to fulfill them . Ofcourse now there are abundance of interactive
WebRTC API each using a new masking method to call the same old WebRTC API function of getusermedia and peer-connection . Few of these I will list down in this blog but no concrete stable reliable guide to setup the backbone network ( yes i am referring to Media inter conversion , relay , TURN , STUN servers ) which is left to telecom software engineer / developer to find out and configure . Instead I see many commercial service providers who claim of providing their backend for our WebRTC implementation but that in my opinion completely defeats the objective of WebRTC based communication . WebRTC was meant to *everything you can’t do with proprietary communication tools and networks* .
Well moving on , here are some nice API implementations of WebRTC ( only for Websockets no SIP )