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Comment by userbinator | Hacker News Reader

On the other hand, there are now many proprietary voice/video communications platforms NOT based on SIP, with some of them adding SIP support only as an afterthought/additional-cost feature. I don't think the authors of SIP envisioned this, and it's unfortunate. It was intended to be as widespread as email, with a similar diversity of implementations of varying interoperability.

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It would have required much forethought to predict, in 2002, that realtime communications would be happening in the browser. Thus WebRTC was born. But a tragedy of WebRTC is that it is not by-default compatible with SIP. WebRTC authors could have specified SIP as the signaling protocol over websocket, but they left the spec open ended. I feel two ways about this: glad for choice, but disappointed that the choice most often made does not allow for interop with the "legacy" realtime communications protocol, SIP.

There are libraries to do SIP signaling over WS/WSS with WebRTC from a browser (or whatever) that theoretically allow for interop with other SIP devices.

In practice the fundamentals of WebRTC rely on things like ICE, STUN, and TURN for media so they're not going to be compatible with almost all existing SIP implementations - many of which can't even do interop with bog-standard vanilla SIP over UDP with standard codecs properly.

As is often the case with SIP you're back to using some Session Border Controller or equivalent architectural component to make interop actually work reliably.

It is a shame. I don't see any easy answer. I am working on sfu-to-sfu[0] and hope it can make some traction. If we can get all the Open Source WebRTC servers working together, maybe there is hope? I believe WebRTC did the right thing. It was as flexible as possible to make it more palatable. You can standardized/codify things, but you can't undo it :)

I am also really excited about WHIP[1]

[0] https://github.com/matrix-org/sfu-to-sfu

[1] https://datatracker.ietf.org/doc/draft-ietf-wish-whip/

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