Update 2014-01-29: RFC7118 has now been officially published. The full specification can be found on the IETF Tools site.
WebRTC developers rejoice! Yesterday, the IETF assigned an RFC number to the SIP WebSocket specification, a drafted document to standardize transporting SIP signaling using HTML5's realtime WebSocket API. The new document, The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP), will be published as RFC7118.
This is big news for WebRTC developers. WebRTC is a free, open source project that enables browsers with Real-Time Communications. When building an application with real time communications, however, one of the first things developers realize is that while WebRTC provides a high quality media stack right in your browser, it does not mandate a certain signaling protocol. This leaves application developers to research many complex protocols or to write their own homegrown signaling.
We here at OnSIP are obviously fans of SIP in general, as using SIP allows us to integrate applications like InstaCall and GetOnSIP directly into the OnSIP Network. We also pride ourselves on being an open, standards-compliant service, which is why we're so excited that OnSIP WebRTC applications will now be entirely based on approved standards.
Furthermore, having a standard specification for sending SIP signaling over WebSockets opens the door for better support from both browsers and servers. Currently, JsSIP and sipML5 are JavaScript SIP stacks that can be used with WebRTC. Many popular SIP proxies, such as Kamailio and OverSIP, as well as soft switches such as FreeSWITCH and Asterisk, already support receiving WebSocket traffic. Once RFC7118 is published, however, look to see more projects popping up with that functionality.
Building a WebRTC application can be daunting, but the signaling shouldn't have to be. RFC7118 is one more step in that direction.