On Fri, Jun 23, 2023 at 11:38 PM TTT <li...@telium.io> wrote: > 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. >
The signaling can go over whatever transport (UDP, Websocket, TCP, TLS). Websockets are commonly used because as I stated in my other response it is what the browser provides. From a media level WebRTC itself is different because it uses additional standards than a regular SIP client. It does ICE, STUN, TURN, DTLS-SRTP (which makes the SDP incompatible with non DTLS-SRTP SDP), and others for media streams, packet loss, and more. Could a normal SIP client use those? Yes. Do they? Usually no. All of this isn't driven by Asterisk, but WebRTC. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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