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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal>I’m learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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 <span style='font-size:10.0pt;font-family:"Courier New";color:#333333'>res_pjsip_transport_websocket, so does that mean Asterisk requires the signaling to occur over a websocket? <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:#333333'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:#333333'>If I used a SIPJS fork which places the signaling over UDP (eg <a href="https://github.com/cwysong85/sipjs-udp">https://github.com/cwysong85/sipjs-udp</a>) will it just be a regular SIP client and I shouldn’t have to configure anything special in Asterisk, just regular PJSIP.</span><span style='font-size:10.5pt;font-family:"Helvetica Neue";color:#333333'><o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>