[asterisk-users] Why is WebRTC treated differently from regular SIP in Asterisk

[TTT]

I'm learning about WebRTC clients, and am wondering why Asterisk treats them
differently from any other SIP client.

 

The media (RTP) should be no different, so the only difference should be on
the signaling side.  I noticed that the Asterisk wiki mentions the need for
res_pjsip_transport_websocket, so does that mean Asterisk requires the
signaling to occur over a websocket?  

 

If I used a SIPJS fork which places the signaling over UDP (eg
https://github.com/cwysong85/sipjs-udp) will it just be a regular SIP client
and I shouldn't have to configure anything special in Asterisk, just regular
PJSIP.

 

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