Short technical dialogue (Q&A) about whether SIP can be used for WebRTC signaling. Speakers state that WebRTC leaves signaling unspecified, that traditional SIP over UDP cannot run in browsers due to lack of raw UDP access, and that SIP must be used over WebSockets (an RFC defines SIP over WebSocket).
Can I use it for WebRTC signaling?
WebRTC doesn't define what you can do for signaling. Or rather it doesn't tell you what to use for signaling. You can decide to do whatever you want. You can use SIP, but you're not going to use SIP over UDP, which is a traditional SIP. You need to do SIP over WebSockets.
Why is that?
Because you don't get the raw UDP access in the browser.
Okay, so if I Want to use WebRTC inside the browser and I need to pick and choose a signaling protocol, I still need to choose a signaling protocol that can run in the browser. And that means running over either HTTPs or websocket.
Yes, which.
Is why we have an RFC that defines now SIP over websocket.
Yes. So it's not pure SIP anymore. It's just a different transport for sip. SIP had to change to be compatible with Rucker C in that respect.
Okay.
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