Displaying 1 result from an estimated 1 matches for "cwysong85".
2023 Jun 24
1
Why is WebRTC treated differently from regular SIP in Asterisk
...nly 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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