SIP is a good protocol to use with WebRTC because SIP already uses SDP for media negotiation, that combined with OnSIP's open and standard SIP architecture makes using SIP for WebRTC a no brainer. The IETF spec for WebRTC specifically does not specify a protocol for use with WebRTC. WebRTC is the media engine that has been implemented in browsers - its interface is SDP, it has nothing to do with SIP, or XMPP, or any other signalling protocol. To clarify some of the info in this thread re: WebRTC.
#Onsip ping test for free
So TLDR have fun hacking around with SIP calling, but if you start trying to use credentials for free PSTN termination, we will shut you down. That said - the bad news is that if OnSIP detects that a client (such as FireRTC/Icehook) is having authorization credentials compromised and used for unauthorized pstn termination, we will immediately disable the users that have been compromised. So I definitely support "playing with SIP stuff" as that is basically what I do for a living I also am very much a SIP enthusiast and maintain an open source implementation of a WebRTC capable federated sip server based on OpenSIPS and RTPEngine - search federated-sip on github (sorry no links, my account is new). Hey guys, I'm Eric, principal engineer at OnSIP.įirst and foremost I want to say that OnSIP is a federated SIP platform, and we very much encourage using standardized DNS based federated SIP.
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